JPH10313498A - Method for picking up sound by sneaking sound suppression, system and recording medium - Google Patents

Abstract

PROBLEM TO BE SOLVED: To sufficiently suppress occurrence of howling with a comparatively simple configuration and insignificant sound quality deterioration. SOLUTION: A microphone 1 is placed to a taker side and a microphone 2 is placed as a speaker driven by a received signal from a subscriber, where output channel signals of the microphones 1, 2 are split into a plurality of bands so that a major component of one frequency band results from one sound source signal (4). An inter-channel level difference/arrival time difference for each identical frequency band are detected and compared with respective threshold levels for each band, and it is determined whether the frequency band is a voice signal component of the talker or other signal component (601). Then the frequency band component of the output of the microphone 1 is selected only for the frequency band discriminated as the voice signal (602) and they are synthesized and the result is transmitted to the subscriber (7A).

Description

DETAILED DESCRIPTION OF THE INVENTION

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention radiates a received signal from the ground as an acoustic signal with a loudspeaker or the like, and collects a voice signal of a speaker with a microphone and transmits the signal to the ground. The converted sound signal is picked up by the microphone, and howling is suppressed.In other words, the sneaking sound suppression type sound collecting method for suppressing the sneaking of the sound signal of the received signal to the signal transmitted to the ground, The present invention relates to an apparatus and a program recording medium thereof.

[0002]

2. Description of the Related Art Conventional methods capable of suppressing unnecessary wraparound noise and suppressing howling are roughly classified into three types. The first method detects a frequency at which howling occurs, and introduces a notch filter of the frequency into a transmission signal (speaker voice signal) or a reception signal. In this method, the sound quality is deteriorated because the band component including the notch filter is missing from the transmission signal.

[0003] The second technique is to receive and reproduce a signal,
In this method, howling is suppressed by performing frequency modulation so that the frequency characteristics of a signal to be collected and transmitted are different. The signal received and reproduced can be reliably grasped as an electric signal. On the other hand, the collected signal includes a signal to be collected and transmitted and a signal to be received and reproduced, and the signal to be collected and transmitted cannot be reliably grasped. Therefore, modulation is applied to signals to be collected and transmitted, which are not necessary for suppressing howling, thereby deteriorating sound quality.

As a third technique, there is a method of predicting, by using an adaptive filter, a situation where a signal received and reproduced is mixed with a signal to be collected and transmitted. Howling is suppressed by subtracting the component of the reproduction signal predicted to be mixed from the signal to be collected and transmitted. However, the adaptive filter for prediction varies every moment, and the prediction of the adaptive filter is possible only when there is no signal to be collected and transmitted, that is, when there is only a reproduced signal. As in the second method, there are many cases where the collected signal is mixed with a signal to be collected and transmitted and a signal to be received and reproduced, and it is sure that there is no signal to be collected and transmitted. Need to be done.

[0005]

Therefore, the prior art has a problem that the deterioration of sound quality is small and howling cannot be suppressed.

[0006]

A sound pickup method according to the present invention comprises:
Using a plurality of microphones provided apart from each other, each output channel signal of each microphone is divided into a plurality of frequency bands in a band division process, and each band mainly includes only one sound source signal component. For each of the same bands of each of the divided output channel signals, the parameters of the acoustic signal reaching the microphone, that is, the level (power) and the arrival time (phase), which change due to the positions of the plurality of microphones, The value difference is detected as a band-specific inter-channel parameter value difference, and the band-divided output channel signal is calculated based on a preset threshold value using the band-specific channel parameter value difference of each band. It is determined in the voice signal determination process whether or not the voice signal component of the speaker, and based on the determination in the voice signal determination process, the band is divided. At least one audio signal input from the same speaker is selected from the power channel signal in the audio signal selection process, and a plurality of band signals selected as signals from the same speaker in the audio signal selection process are output as the audio signal. In the speech synthesis process, and transmits the synthesized speech signal to the ground.

According to an embodiment of the present invention, a signal received from the ground is divided into a plurality of bands as narrow as a band having only a negligibly small level in one band, and the band is divided. The level of each received signal is detected, and for each of these divided bands,
If the detected level is equal to or less than a predetermined value, the transmission band is determined in the transmission band determination process, and only the band determined to be transmittable in the band signal selected in the audio signal selection process is selected. And send it to the speech synthesis process.

[0008] The selection in the transmission possible selection step may be performed by performing the determination in the audio signal determination step only on the band determined to be transmittable. According to another embodiment of the present invention, a received signal is divided into a plurality of frequency bands, and a band-divided received signal component corresponding to the band selected in the audio signal selecting step is subjected to a frequency component removing step. The band component of the remaining received signal from which the component has been removed is
The signal is synthesized with the signal in the time domain in a re-synthesis process, and the synthesized signal is supplied to the electroacoustic conversion means.

[0009]

DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS FIG. 1 shows a basic configuration used in a sound pickup apparatus according to the present invention. In FIG. 1, a speaker 211 is provided as an electroacoustic conversion unit in a room 210, and a transmission line 212 is provided.
Signal (received signal) of the other party's speaker sent via
Is reproduced by the speaker 211 and emitted into the room 210 as an acoustic signal. On the other hand, a voice signal uttered by the speaker 215 in the room 210 is received by the microphone 1 and transmitted as an electric signal to the partner speaker through the transmission line 216.
In this case, howling occurs when an audio signal emitted from the speaker 211 is captured by the microphone 1 and transmitted to the other party.

Therefore, in this embodiment, the microphone 2 is provided in parallel with the arrangement direction of the speaker 211 and the speaker 215, and the microphone 1 is provided, for example, about 20 cm apart, and the microphone 2 is on the speaker 211 side. These microphones 1 and 2 are connected to the sound collection processing unit 220. FIG. 2 illustrates a specific example of the sound collection processing unit 220. The output of the microphone 1 is called an L channel signal, and the output of the microphone 2 is called an R channel signal. The L-channel signal and the R-channel signal are supplied to an inter-channel time difference / level difference detection unit 3 and a band division unit 4, where they are divided into a plurality of frequency band signals, and the inter-channel time difference / level difference for each band. The signal is supplied to the detection unit 5 and the sound source determination signal selection unit 6. According to each detection output of the detection units 3 and 5, the selection unit 6 selects any channel signal for each band as a voice component of a speaker or a sound component from a speaker, and the speaker for each of the selected bands. The voice component signals are synthesized by the voice signal synthesis unit 7A, and only the speaker voice signal is extracted.

Since the speaker 215 is closer to the microphone 1 than the microphone 2, the speaker's voice reaches the microphone 1 earlier and has a higher level, and the speaker 211 is connected to the microphone 2 more than the microphone 1.
, The acoustic signal from the speaker 211 reaches the microphone 2 earlier than the microphone 1 and has a higher level. As described above, according to the present invention, the amount of change in the acoustic signal reaching the microphones 1 and 2 due to the positions of the speaker as the sound source and the speaker with respect to the microphones 1 and 2, in this example, the arrival time difference and the level difference between the signals. Use.

In the audio signal determination unit 201, a threshold value for each band, for example, a difference between the level difference and the arrival time is 0 for comparison.
If it is larger, the component of the band is determined to be the speaker's voice component, and if smaller than 0, the component of the band is determined to be the speaker acoustic component. However, this is a case where the level and the arrival time obtained from the output signal of the microphone 2 are subtracted from the level and the arrival time obtained from the output signal of the microphone 1 by the difference detection unit 5.

The audio signal selecting unit 602 selects the band components of the signal of the microphone 1 only for the band determined to be the speaker's voice in this way, and the selected band audio component is converted by the audio signal synthesizing unit 7A. The signal is converted into a time domain signal, that is, a synthesized voice signal, and transmitted to the transmission path 216. Hereinafter, an example of a method of extracting a speaker voice signal separately from a speaker acoustic signal and extracting the same will be specifically described. In the following, speaker 21
5 and the speaker 211 are referred to as sound sources A and B, respectively. For example, when there are a plurality of speakers, each audio signal of these speakers is separated and only one or a plurality of them are transmitted. Because.

[0014] As shown in FIG.
Signals from the two sound sources A and B are taken in (S0
1). The inter-channel time difference / level difference detection unit 3
Time difference between channels from the channel signal and the R channel signal or
Detect level differences. Parameters used to detect time differences
Is the mutual phase between the L channel signal and the R channel signal.
The case where a function is used will be described. As shown in FIG.
Of each sample of the L channel signal and the R channel signal
(T) and R (t) are read (S02),
Then, a cross-correlation function between the files is calculated (S03). This calculation is
Cross-correlation of both channel signals at the same sample time
And one channel signal for the other channel
If the signal is shifted one sample time, only two
Cross-correlation in each case of shifting ...
To find the cross-correlation function. Find a large number of these cross-correlations,
Create a histogram that normalizes these with power
(S04). Next, the cumulative frequency ranking of the histogram
Time difference Δα to take second place and second place respectively ₁, Δα_TwoAsk for
(S05). These time differences Δα₁, Δα_TwoIs given by
Time difference between channels Δτ₁, Δτ_TwoConversion to
And output (S06).

Δτ ₁ = 1000 × Δα ₁ / F (1) Δτ ₂ = 1000 × Δα ₂ / F (2) where F is a sampling frequency, and setting it to 1000 times is to increase the value to some extent for the sake of computational convenience. It is. The time differences Δτ ₁ and Δτ ₂ are the time differences between the channels of the L channel signal and the R channel signal of the signals of the sound sources A and B, respectively.

Returning to the description of FIGS.
The channel signal and the R channel signal are divided into signals L (f1), L (f2),..., (Fn) and signals R (f1), R (f2),. S
04). This division is performed, for example, by discretely Fourier-transforming each channel signal into a frequency-domain signal, and then dividing it into frequency bands. In this band division, in the case of an audio signal, for example, a 20 Hz bandwidth is used so that only the signal component of one of the sound sources mainly exists in each band from the difference in the frequency characteristics of the signals of the sound sources A and B. The power spectrum of the sound source A is obtained, for example, as shown in FIG. 5A, and the power spectrum of the sound source B is obtained as shown in FIG. 5B, and each of the spectra is divided by a bandwidth Δf that can be separated. At this time, the spectrum of one sound source can be neglected with respect to the spectrum of the other sound source, for example, as shown by the corresponding spectrum with a broken line. FIG. 5
As may be understood from A and 5B, they may be separated by a bandwidth 2Δf. That is, it is not necessary to include only one spectrum in each band. The discrete Fourier transform is performed, for example, every 20 to 40 ms.

Next, for example, L (f1), R (f1),... L (f)
A time difference or a level difference between channels for each band is detected between the channels of the corresponding band signals such as n) and R (fn) (S05). Here, the inter-channel time difference for each band is uniquely detected by using the inter-channel time differences Δτ ₁ and Δτ ₂ detected by the inter-channel time difference detection unit 3. The equation used for this detection is as follows.

Δτ₁− {(Δφi / (2πfi) + (ki1 / fi)} = ε_i1 (3) Δτ_Two− {(Δφi / (2πfi) + (ki2 / fi)} = ε_i2 (4) i = 1, 2,..., N, Δφi are the signal L (fi) and the signal R
(Fi) is the phase difference. In these equations, ε_i1, ε_i2
The integers ki1 and ki2 are determined so as to be minimum. next,
Its minimum ε_i1 and ε_iCha that is smaller than 2
Flannel time difference Δτ _j(J = 1, 2) is replaced by the channel i
Time difference between tunnels Δτ_ijAnd That is, one of the sound source signals
Is the time difference between channels in the band.

The sound source determination signal selection section 6 uses the band-to-channel time differences Δτ _{1j to} τ _nj detected by the band-to-channel time difference / level difference detection section 5 to generate each band signal L (f1).
To L (fn) and R (f1) to R (fn) to determine which one to select, the audio signal determination unit 6
01 (S06). For example, of the time differences Δτ ₁ and Δτ ₂ calculated by the inter-channel time difference / level difference detection unit 3, Δτ ₁ is the inter-channel time difference of the signal from the sound source A close to the L-side microphone, and Δτ ₂ is , And the time difference between channels of the signal from the sound source B, which is close to the microphone on the R side.

In this case, the band i in which the time difference Δτ _ij calculated by the band-by-band channel time difference / level difference detection unit 5 is Δτ ₁ is determined by the audio signal determination unit 601 by the gate 602L.
i is opened and the input signal L (fi) on the L side remains at S
A (fi), and the input signal R of the band i on the R side
(Fi) indicates that the audio signal determination unit 601 uses the gate 602R.
Is closed, and SB (fi) is output as 0. The band i in which the time difference Δτ _ij becomes Δτ ₂ is, on the contrary, the signal L on the L side.
(Fi) is output as SA (fi) = 0, and on the R side, the input signal R (fi) is output as it is as SB (fi). That is, as shown in FIG. 1, the band signals L (f1) to L (f1) to L
(Fn) are supplied to the sound source signal synthesis unit 7A through the gates 602L1 to 602Ln, respectively, and the band signal R (f
1) to R (fn) are gates 602R1 to 602, respectively.
The signal is supplied to the sound source signal synthesis unit 7 through Rn. In the sound signal determination section 601 in the sound source determination signal selection section 6, Δτ _1j -Δ
For the band i for which τ _nj is input and Δτ _ij is determined to be Δτ ₁ , gate control signals CLi = 1 and CRi = 0 are generated, and the corresponding gate 602Li is controlled to be open and 602Ri is controlled to be closed, and Δτ _{ij is} controlled. Is determined to be Δτ ₂ , the gate control signal CLi = 0 and CRi = 1
Is generated and the corresponding gate 602Li is closed, 602R
i is respectively controlled to be open. The above description is a functional configuration, and is actually processed by, for example, a digital signal processor.

The signals SA (fi) to
SA (fn) are combined, the band in the example of the division is the inverse Fourier transform respectively, is output to the output terminal t _A as a signal SA, and the signal SB by the sound source signal synthesizer 7B (fi)
To SB (fn) is output to the output terminal t _B as similarly synthesized by the signal SB. As is clear from the above explanation,
In the apparatus of the present invention, it is determined from which sound source each band component is obtained by finely band-dividing each channel signal, and all the determined components are output, that is, the signals of the sound sources A and B are output. If the frequency components do not overlap each other, processing is performed without dropping a specific frequency band, so that the signals of the sound sources A and B are separated while maintaining high sound quality as compared with the conventional method of extracting only the harmonic structure. It is possible.

In the above description, the sound source is generated using only the inter-channel time difference and the inter-channel time difference detected by the inter-channel time difference / level difference detection unit 3 and the inter-channel time difference / level difference detection unit 5. The determination condition is determined by the determination signal unit 601. Next, an embodiment in which the determination of the determination condition is processed using the level difference between the channels will be described. In this embodiment, as shown in FIG. 6, an L channel signal and an R channel signal are fetched from the microphones 1 and 2 (S02), and a level difference ΔL between the L channel signal and the R channel signal is detected by a time difference / level difference between channels. The detection is performed by the unit 3 (FIG. 2) (S03). As in step S04 in FIG. 3, the L channel signal and the R channel signal are respectively divided into n band-specific channel signals L (f1) to L (fn), R (f).
1) to R (fn) (S04), and band-specific channel signals L (f1) to L (fn) and R (f1) to R (fn).
, That is, L (f1) and R (f1), L (f
2) and R (f2),..., L (fn) and R (fn), the level difference between channels ΔL1, ΔL2,.
n is detected (S05).

A human voice can be regarded as a steady state for about 20 ms to 40 ms. Therefore, in the audio signal determination unit 601 (FIG. 2), 20 ms to 40 m
For each s, the sign of the value obtained by taking the logarithm of the inter-channel level difference ΔL and the sign of the value obtained by taking the logarithm of the inter-channel level difference ΔLi are the same sign in more than a few percent of the entire band. (+ Or-) is calculated, and a predetermined value, for example, 8
If both have the same code in a band equal to or greater than a certain percentage (S06, S0
7) From 20 ms to 40 ms therefrom, it is determined only by the level difference ΔL between the channels (S08), and if the band having the same code is 80% or less, 20 ms to 40 m from there.
During s, determination is made for each band using the band-based channel level difference ΔLi (S09). When the entire band is determined by the level difference ΔL between channels, if ΔL is positive, the L channel signal L (t) is output as it is as the signal SA, and the R channel signal R (t) is It is output as SB = 0. If ΔL is equal to or less than 0, the L-channel signal L (t) is output as the signal SA = 0, and the R-channel signal R (t) is output as the signal SB as it is. However, this is an explanation in the case where a value obtained by subtracting the R side from the L side is used as the level difference between channels. Further, when the determination is made for each band using the level difference ΔLi between the bands, if the level difference ΔLi between the channels is positive for each band fi, the L-side divided signal L (fi) is directly used as the signal SA. (Fi) and the R-side divided signal R
(Fi) is output as the signal SB (fi) = 0. On the other hand, if the level difference ΔLi is 0 or less, the L side
(Fi) is output as a signal SA (fi) = 0, and on the R side, a divided signal R (fi) is output as a signal SB (fi). As described above, the gate control signals CL1 to CLn and CR1 to CRn are output from the audio signal determination unit 601 and the gates 602L1 to 602Ln, 602R1 to 60
2Rn are respectively controlled. This is also a case where a value obtained by subtracting the R side from the L side is used as the level difference between channels for each band, as in the former case. Signals SA (f1) to SA
(Fn), the signal SB (f1) ~SB (fn) signals are respectively similar to the previous embodiments the synthetic SA, an output terminal t _A as SB, are output to t _B (S10).

In the above embodiment, only one of the arrival time difference and the level difference is used as the judgment condition used in the audio signal judgment unit 601. However, if only the level difference is used, L (fi) and R
In some cases, the level with (fi) may antagonize, in which case it is difficult to accurately determine the level difference. Also, when only the time difference is used, in a high frequency band,
In some cases, it is difficult to calculate the time difference correctly due to the rotation of the phase. From these points, it is better to use the time difference for the low frequency band and the level difference for the high frequency band,
It may be advantageous to use a single parameter over the entire band.

An embodiment in which the audio signal determination section 601 uses both the time difference between channels for each band and the level difference between channels for each band will be described with reference to FIGS. The blocks of the functional configuration of this embodiment are the same as those shown in FIG. 2, but an inter-channel time difference / level difference detecting section 3, an inter-channel time difference / level difference detecting section 5, and an audio signal judging section 6 are provided.
01 differs as follows. The inter-channel time difference / level difference detector 3 detects the detected time differences Δτ ₁ , Δτ
If the average of the absolute values of ₂ or Δτ ₁ and Δτ ₂ are relatively close values, one time difference Δτ is output, such as only one of them. Although the inter-channel time differences Δτ ₁ , Δτ ₂ , and Δτ have been calculated before band division of the channel signals L (t) and R (t) on the frequency axis, they may be calculated after band division.

As shown in FIG. 6, the L channel signal L
(T), the R channel signal R (t) is converted to a frame (for example, 2
0 to 40 ms) (S02), and the band dividing unit 4
Divides the L channel signal and the R channel signal into a plurality of frequency bands, respectively. In this example, the L channel signal L
(T), a Hanning window is applied to each of the R channel signals R (t) (S03), and the signals L (f1) to L (fn) and R (f1) to R divided by applying the Fourier transform are respectively applied.
(Fn) is obtained (S04). Next, the band-by-band time difference between channels / level difference detector 5 detects the frequency f of the divided signal.
It is checked whether or not i is a band (hereinafter, referred to as a low band) equal to or less than 1 / (2Δτ) (Δτ is a channel time difference) (S05), and if it is, a phase difference Δφi between bands is output (S05).
08), it is checked whether or not the frequency f of the divided signal is a band larger than 1 / (2Δτ) and smaller than 1 / Δτ (hereinafter referred to as a middle band) (S06). The inter-channel phase difference Δφi and the level difference ΔLi are output (S09), and it is checked whether the frequency f of the divided signal is equal to or more than 1 / Δτ (hereinafter, referred to as high band) (S0).
7) If the frequency is in the high frequency range, the inter-channel level difference ΔLi is output (S10).

The audio signal determination unit 601 uses L (f1) to L (f) using the band-to-channel phase difference and level difference detected by the band-to-channel time difference / level difference detection unit 5.
n), and which of R (f1) to R (fn) is output is determined. In this example, values calculated by subtracting the value on the R side from the L side are used for the phase difference Δφi and the level difference ΔL.

The signals L (fi) and R (f
Regarding i), first, as shown in FIG.
It is checked whether it is the above (S15).
The value obtained by subtracting π is set to Δφi (S17), and step S1
If Δφi is not equal to or more than π in 5, it is checked whether it is equal to or less than −π (S
16), the value obtained by adding 2π to Δφi is Δφ
i (S18), and if it is not less than -π in step S16, Δφi is used as it is (S19). Step S1
7, phase difference Δ between channels obtained in band obtained in S18, S19
φi is converted into a time difference Δσi by the following equation (S20).

Δσi = 1000 · Δφi / 2πfi (5) When the divided signals L (fi) and R (fi) are determined to be in the middle band, as shown in FIG. ) Is used to uniquely determine the phase difference Δφi. That is, it is checked whether ΔL (fi) is positive (S23). If positive, it is checked whether the phase difference Δφi for each band is positive (S24). If positive, the Δφi is output as it is (S26). ), Δφi if not positive in step S24
Is output as Δφi (S2
7). If ΔL (fi) is not positive in step S23,
It is checked whether the band-by-band phase difference Δφi is negative (S25). If negative, the Δφi is directly used as Δφi
(S28), and a value obtained by subtracting 2π from Δφi is output as Δφi unless it is negative in step S25 (S29). Any of these Δφi in steps S26 to S29 is represented by
Is calculated (S30).

Δσi = 1000 · Δφi / 2πfi (6) As described above, the inter-channel time difference Δσi in the low band and the middle band and the inter-channel level difference ΔL (fi) in the high band are obtained. Is determined in the following manner. As shown in FIG. 10, using the phase difference Δφi in the low band and the middle band, and using the level difference ΔLi in the high band, each frequency component of both channels is determined as a signal of one of the corresponding sound sources. Specifically, it is checked whether or not the band-by-band channel time difference Δσi obtained in FIGS. 8 and 9 is positive in the low band and the middle band (S3).
4) If positive, the L-side channel signal L of the band i
(Fi) is output as a signal SA (fi), and the R-side band channel signal R (fi) is output as a signal SB (fi) of 0 (S36). If the band-based channel time difference Δσi is not positive in step S34, on the contrary, SA (fi) is set to 0.
And outputs the R-side channel signal R (f) as SB (fi).
i) is output (S37).

In the high frequency range, the channel-to-channel level difference ΔL (f) detected in step S10 in FIG.
It is checked whether i) is positive (S35).
The L-side channel signal L (fi) is output as A (fi), and 0 is output as SB (fi) (S38). If the level difference ΔLi is not positive in step S35, SA (f
It outputs 0 as i) and outputs the R-side band channel signal R (fi) as SB (fi) (S39).

As described above, the L side or the R side is output for each band, and the frequency components determined by the sound source signal combining units 7A and 7B are added over the entire band (S4).
0) and inversely Fourier-transform each added signal (S4
1) Output the converted signals SA and SB (S4)
2). As described above, in this embodiment, by using parameters that are advantageous for sound source separation for each frequency band, sound source separation with higher separation performance is realized as compared with the case where a single parameter is used over the entire band. It is possible to

The present invention is applicable even when the number of sound sources is three or more.
Wear. For example, if the number of sound sources is 3 and the number of microphones is 2,
Source time difference using the arrival time difference to the microphone
The case of separation will be described. In this case, the time difference between channels /
In the level difference detection unit 3, L channel signal, R
When calculating the time difference between channels of the channel signal, FIG.
Histogram normalized by cross-correlation power as shown in
The ram's cumulative frequency (peak value)
By determining each time point taken,
Time difference between channels Δτ₁, Δτ_Two, Δτ _ThreeIs calculated.
Then, the band-by-band time difference between channels / level difference detection unit 5
In addition, the time difference between channels in each band is Δτ₁Or
Δτ_ThreeDecide on one of How to determine this
This is the same as the calculation formulas (3) and (4) described in the embodiment. sound
In the voice signal determination unit 601, for example, Δτ₁> 0, Δτ
_Two> 0, Δτ_ThreeThe case where <0 is described. Where Δ
τ₁, Δτ_Two, Δτ_ThreeAre the signals of sound sources A, B, and C, respectively.
Signal time difference between channels, and furthermore, these values are
It is assumed that the value is calculated by subtracting the value on the R side from the L side. this
In this case, the sound source A is close to the microphone 1 on the L side, and the sound source B is
It is near the microphone 2 on the R side. Therefore, L channel
Time difference between channels for each band is Δτ₁Becomes
The signals of the sound source A are added by adding the signals of the bands, and Δτ_TwoTona
And separate the signals of sound source B from each other.
And it is possible. Also, from the R channel signal,
The time difference between channels is Δτ_ThreeSignal of the band
By applying the force, the signal of the sound source C is separated.

In the above sound source separation, the speaker 215
And the speaker 211 are fixed, the time difference Δτ ₁ (or Δτ ₂ ) at which the sound signal from the speaker 215 (or the speaker 211) reaches the microphones 1 and 2 is constant, and it is known in advance. And the inter-channel level difference ΔL can be known in advance. Therefore, the detection of the inter-channel time differences Δτ ₁ and Δτ ₂ in step S03 in FIG. 3 and the detection of the inter-channel level difference ΔL in step S03 in FIG. 6 can be omitted, and the inter-channel time difference / level in FIG. The difference detector 3 can be omitted. In the case of using the band-based channel level difference ΔL (fi), in FIG. 6, steps S03, S06, S07, and S08 are omitted, and the band-based channel level difference is always used for each divided band. Sound source separation may be performed. That is, the level difference between the channels need not be detected. However, if p / n ≧ 0.8 is satisfied by performing the processing shown in FIG. 6, the processing is simplified.

In the above description, the sound source signals are separated, and the separated sound source signals SA and SB are separately output. However, for example, one sound source A is a voice by a speaker and the other sound source B
In the case of a noise, the present invention can also be applied to separate and extract the signal sound of the sound source A mixed with the noise and suppress the noise. When one sound source A, for example, the speaker has a wider frequency band than the other sound source B, that is, the speaker, and the respective frequency bands are known in advance, as shown in FIG. In addition, a frequency band in which both sound source signals do not overlap is separated. For example, the frequency band of the signal A (t) of the sound source A is f1 to fn, while the frequency band of the signal B (t) of the sound source B is f1 to fn (f
n> fm), non-overlapping bands fm + 1 to fn
Are separated from the outputs of the microphones 1 and 2, and the signals of the bands fm + 1 to fn are separated by the audio signal determination unit 6.
01, and in some cases, the processing of the band-by-band time difference / level difference detection unit 5 is not performed, and the audio signal determination unit 601 selects the channel signal S selected as the signal of the sound source B.
The divided band channel signals R (fm + 1) to R (fn) of R to be selected as B (t) are respectively expressed as SB (fm +
1) to SB (fn) and output as SA (fm + 1) to
SA (fn) outputs the audio signal selecting unit 6 so that 0 is output.
02 is controlled. That is, the gate 602Lm + 1 to 602L
n is normally closed, and gates 602Rm + 1 to 602Rn are normally open.

In the above description, the time difference between channels Δσ for each band
i, positive or negative, and whether the level difference ΔLi between the respective channels is positive or negative, that is, 0 is set as the threshold value, and it is determined which microphone is close to the band signal. This is a case where the sound source A and the sound source B are located symmetrically with respect to the bisector of the line connected as the microphone 1. If not, the determination threshold value may be determined as follows.

The level difference between the channels at which the signal of the sound source A reaches the microphones 1 and 2 is represented by ΔL _A ,
Arriving per-band channel between the time difference .DELTA..tau _A, signal microphone 1 of the sound source B, [Delta] L _B the level difference between the band-by-band channel that reaches the microphone 2, respectively and .DELTA..tau _B the time difference between the arriving band-by-band channel. In this case, the threshold DerutaLth the per-band channel level difference is set to _{ΔLth = (ΔL A + ΔLi)} / 2, the threshold Derutatauth the per-band channel between the time difference if _{Δτth = (Δτ A + Δτ B} ) / 2 Good. In the embodiment described above, ΔL _B = −Δ
When L _A , Δτ _B = −Δτ _A , ΔLth = 0, Δτth =
It becomes 0. In order to separate the sound sources A and B, the microphones 1 and 2 are positioned so that the two sound sources are on the different sides with respect to the microphones 1 and 2, and the distances and directions to the microphones 1 and 2 are not always known correctly. In some cases, the threshold values ΔLth and Δτth may be variable, and the threshold values ΔLth and Δτth may be adjustable so that separation is performed well.

FIG. 12 shows a further improvement of the howling suppressing method. A branching unit 231 is inserted into the transmission line 212 from the other side connected to the speaker 211, and the audio signal from the other side speaker, which is branched by this, is delayed by the delay unit 232 as necessary, and then divided by the band division unit. At 233, it is divided into a plurality of frequency bands. This division is equal to the number of divisions performed by the band division unit 4 and may be performed by a similar method. The component of each band of the audio signal that has been band-divided by the other party is analyzed by the transmittable band determining unit 234, and it is determined whether the frequency band of the component is a transmittable frequency band. That is, a band having no frequency component of the audio signal from the other party or a band having a sufficiently low level is determined as a transmittable band. The dividing unit 4 also divides the received signal from the other party into a band narrow enough to obtain a band in which the components of the received signal can be ignored.

A transmittable component selector 235 is inserted between the audio signal selector 602L and the sound source signal synthesizer 7A.
The audio signal selection unit 602L determines and selects the output signal S1 of the microphone 1 as the audio signal of the speaker 215,
Further, only the band components determined to be transmittable by the transmittable band determination unit 234 are selected by the transmittable component selection unit 235 and sent to the sound source signal combining unit 7A. Therefore, the frequency component that is output from the speaker 211 and may cause howling is not transmitted to the transmission line 216, and the occurrence of howling can be suppressed more reliably. As the transmittable component selection unit 235, the audio signal determination unit 601 may determine only the band that is determined to be transmittable by the transmittable band determination unit 234, and may not transmit other bands.

The delay unit 232 determines the amount of delay in consideration of the propagation time of the acoustic signal between the speaker 211 and the microphones 1 and 2. Means for obtaining the delay effect performed by the delay unit 232 include a branching unit 231 and a transmittable component selection unit 2
35 may be inserted after any processing stage. When the data is inserted after the transmittable band determination unit 234 as shown by a dotted frame 237, a readable and writable recording unit that accumulates data is used, and after a time corresponding to the required delay amount,
The information can be read and supplied to the transmittable component selection unit 235. In short, it is sufficient to provide a means for delaying the control of the transmittable component selection unit 235 in consideration of the propagation time. In some cases, these delay means can be omitted.

In the embodiment shown in FIG. 12, a component which may cause howling is cut off on the transmission side (output side), but may be cut off on the reception side (input side). FIG. 13 shows a main part of the embodiment. The received signal from the transmission line 212 is transmitted to the band division unit 241.
Is divided into a plurality of frequency bands. This division is the same as that of the band division unit 4 (FIG. 2), and can be performed by the same method. This band-divided received signal is input to frequency component removing section 242. A control signal obtained from the audio signal determination unit 601 and used by the audio signal selection unit 602L to select the audio component of the speaker 215 from the microphone 1 is input to the frequency component removal unit 242, and the audio signal selection unit 602
A band component not selected by L, that is, a band component not transmitted to the transmission line 216 is selected from the reception signal band-divided by the frequency component removal unit 242 and supplied to the acoustic signal synthesis unit 243,
The two are combined into an acoustic signal and supplied to the speaker 211. The sound signal synthesizing section 243 has the same function as the sound source signal synthesizing section 7A. According to this configuration, the speaker 2
Since the frequency component transmitted to the transmission line 216 is excluded from the acoustic signal emitted from the audio signal 11, the occurrence of howling is suppressed.

As described in the embodiment of FIG. 2, the thresholds ΔLth and Δτth for determining to which sound source signal the band component belongs from the time difference between channels for each band and the level difference between channels for each band. Preferred values differ depending on the relative positions of the microphone and the microphone. Therefore, the threshold value setting unit 251 is provided as shown in FIG.
01, that is, the threshold values ΔLth, Δτth
Is preferably changed according to the situation.

Further, in order to improve noise resistance, a reference value setting unit 252 is provided to set a silence criterion for silencing frequency components having a level equal to or lower than a certain value, and send the same to the audio signal selection unit 602L. . As a result, in the audio signal selection unit 602L, of the frequency components of the collected sound signal of the microphone 1 selected based on the level difference threshold value and the phase difference (time difference) threshold value, the frequency component whose level is equal to or less than the predetermined value is dark. It is regarded as noise components such as noise and air-conditioning noise and is removed, thereby improving noise resistance.

By the way, in order to prevent the occurrence of howling, a howling prevention criterion for holding a frequency component having a level equal to or higher than a predetermined value to the reference value setting section 252 is added to the reference value setting section 252. send. As a result,
In the audio signal selection unit 602L, from the frequency difference of the level difference threshold value and the phase difference threshold value, or the frequency component of the collected sound signal of the microphone 1 selected based on the silence criterion added thereto, the level of which is equal to or more than a certain value The frequency component is corrected to a level equal to or lower than the fixed value. This correction is performed by instantaneously and occasionally clipping to the fixed value level when the level becomes higher than the predetermined value level, and by compressing the dynamic range when the frequency becomes relatively higher than the predetermined value level. By doing so, it is possible to suppress an increase in the amount of acoustic coupling that causes howling, and to prevent howling.

As shown in FIG. 13, a configuration for suppressing reverberation can be added. In other words the output terminal t _A, wraparound estimates the echo signal obtained by delaying the signal estimator 26
1 and an estimated wraparound signal subtraction section 262 for reducing the estimated delayed wraparound signal, and utilizing the property of the transfer characteristic between the direct sound and the reverberation sound, to make use of the wraparound signal estimation section 261.
The wraparound signal delayed in is estimated and extracted. For this estimation processing, for example, a complex cepstrum method considering the minimum phase characteristic of the transfer characteristic is used. If necessary, the transfer characteristics between the direct sound and the reverberant sound can be measured by the impulse response method. The echo signal delayed estimated by the estimation unit 261, and sends it to the echo signal removing unit 262 at the output terminal t separated sound source signal from the _A (speaker 215 of the speech signal) transmission line 216 is removed from the. Suppression of a sneak signal by the sneak signal estimation unit 261 and the sneak signal removal unit 262 is described in, for example, the literature, Showa 62
It is shown in "Digital Signal Processing", translated by Date Gen on November 25, 2008 by Corona Co., Ltd. The wraparound signal estimation unit 261 and the wraparound signal removal unit 262 can be processed by, for example, one DSP (digital signal processor).

When the speaker 215 moves only within a certain range, the microphone 1 installed on the side of the speaker 215
And the phase difference / arrival time difference between the frequency component of the sound picked up by the microphone and the frequency component of the sound picked up by the microphone 2 installed on the side of the speaker 211 are limited to a certain range. . Therefore, a determination reference range is set in the threshold value setting unit 251 and signal processing is performed only on the phase difference range within the level difference range, and the signal processing outside the range is not processed. In this way, the uttered voice of the speaker 215 can be selected from the collected signals of the microphone 1 with higher accuracy.

From another viewpoint, the speaker 211 is fixed, so that the frequency component of the sound of the speaker 211 picked up by the microphone 1 of the speaker 215 and the frequency component of the sound of the speaker 211 The level difference, phase difference, or arrival time difference from the frequency component of the sound of the speaker 211 collected by the microphone 2 is limited to a certain range.
The range of the level difference and the phase difference / arrival time difference is also a criterion for discarding in the audio signal selection unit 602L, and a determination criterion for making a selection in the audio signal selection unit 602L is based on the threshold value. It can also be set in the setting unit 251.

Also in this howling suppression, if three or more microphones are used, the function of selecting a necessary frequency component can be achieved with higher accuracy. Further, the present invention has been applied to the sneaking sound suppressing type sound pickup device of a loudspeaker type sound system, but can also be applied to a general telephone transmission / reception device. Also, the audio signal selection unit 6
The frequency component to be selected at 02L is the microphone 1
The frequency component is not limited to a specific frequency component (speech of the speaker 215) among the frequency components of the audio signal picked up in the above. If, for example, the speaker 215 has an air outlet of the air conditioner, A frequency component determined as the voice of the speaker 215 from the frequency components picked up by the microphone 2 is selected, or, in an environment where the noise is large, the speaker 215 among the frequency components picked up by the microphones 1 and 2 is selected. It is also possible to select the frequency component determined to be the voice of the sound.

It has been described above that the present invention can be applied to a case where a plurality of speakers are separated and one or two voice signals are transmitted while suppressing the looping sound. In this case, the synthesized speech signals of a plurality of speakers are obtained separately from each other. However, if the synthesized speech signals corresponding to the speakers who are not speaking are suppressed or cut off, the quality of the transmission speech signal is further improved. For this purpose, it is detected whether or not the speaker is speaking, but it is detected which sound source is not sounding, and a suppression signal for the corresponding synthesized speech signal is created. A method of creating the suppression signal will be briefly described.

As shown in FIG. 14, microphones M1,
M2 and M3 are arranged, for example, at the vertices of an equilateral triangle having one side of 20 cm. A space is divided and set based on the directional characteristics of the microphones M1 to M3, and each divided space is called a sound source zone. All microphones M
When 1 to M3 are omnidirectional and have the same characteristics, for example, as shown in FIG. 12, they are divided into six zones Z1 to Z6. That is, each microphone M1, M2, M3
And six zones Z1 divided into six at equal angular intervals around the center point Cp by straight lines passing through the center point Cp.
To Z6 are formed. Sound source A is located in zone Z3, and sound source B is located in zone Z4. That is, the microphones M1 to M3 are set so that one sound source zone belongs to one sound source zone.
Each sound source zone is determined based on the arrangement and characteristics of the sound source.

In FIG. 14, a band dividing unit 41 divides a sound signal S1 of the first channel collected by the microphone M1 into n frequency band signals S1 (f1) to S1 (fn). The acoustic signal S2 of the second channel collected by the microphone M2 is converted into n frequency band signals S2 (f
1) to S2 (fn), and the band dividing unit 43 divides the sound signal S3 of the third channel collected by the microphone M3 into n frequency band signals S3 (f1) to S3 (fn). These bands f1 to fn are divided into band division units 41 to 4
3, and such band division can utilize a discrete Fourier transformer.

The sound source separation section 80 separates a sound source signal by using the method described with reference to FIGS. However, in FIG. 14, since there are three microphones,
Similar processing is performed for each two combinations of these three channel signals. Therefore, the band division unit and the band division units 41 to 43 in the sound source separation unit 80 can also be used. The level (power) detection section S1 detects the level (power) signals P (S1f1) to P (S1fn) of the signals S1 (f1) to S1 (fn) of each band obtained by the band division section 41, Similarly, each band signal S2 (f) obtained by the band division units 42 and 43 by the band-specific level detection units 52 and 53, respectively.
1) -S2 (fn), S3 (f1) -S3 (fn) P (S2f1) -P (S2fn), P (S3f1) -P
(S3fn) are respectively detected. These band-specific level detections can also be realized by a Fourier transformer. That is, each channel signal may be decomposed into a spectrum by a discrete Fourier transform, and the power of each spectrum may be obtained. Therefore,
A power spectrum may be obtained for each channel signal, and the power spectrum may be divided into bands. Each channel signal of each of the microphones M1 to M3 is divided into each band by the band-specific level detection unit 400, and the level (power) is output.

On the other hand, the level (power) P (S1) of all frequency components of the sound signal S1 of the first channel picked up by the microphone M1 is detected by the all-band level detector 61, and the entire-band level detectors 62 and 63 are detected. With each microphone M
2 and 3, the second and third channels collected by M3, respectively.
Level P of all frequency components of each of the acoustic signals S2 and S3 of FIG.
(S2) and P (S3) are detected.

The sound source state determination unit 70 determines, by computer processing, a sound source zone that does not emit sound.
First, the band-specific levels P (S1f1) to P (S1fn) and P (S2f1) obtained by the band-specific level detection unit 50.
~ P (S2fn), P (S3f1) ~ P (S3fn)
Are compared with each other for signals in the same band. Then, the highest level channel is specified for each of the bands f1 to fn.

By setting the number n of band divisions to a predetermined number or more, as described above, one band can be regarded as containing only the sound signal of one sound source. fi level P (S1fi), P (S
2fi), P (S3fi can be regarded as the level of sound from the same sound source, so that the levels P (S1fi), P (S2) of the same band for the first to third channels.
When there is a difference between fi) and P (S3fi), the level of the band of the microphone channel closest to the sound source becomes the highest.

As a result of the above processing, the highest level channel is assigned to each of the bands f1 to fn. For each of the first to third channels in the n bands,
The total number # 1, # 2, and # 3 of the bands having the highest level is calculated. The microphone of a channel having a larger value of the total number can be regarded as being closer to the sound source. If the total numerical value is, for example, about 90 n / 100 or more, it can be determined that the sound source is close to the microphone of that channel. However, the total number of bands having the highest level is 53n / 10
If the next largest sum is 0n / 100, it is not clear whether the sound source is close to the corresponding microphone. Therefore, the total number is equal to the preset reference value ThP,
For example, when it exceeds about n / 3, it is determined that the sound source is closest to the microphone of the channel corresponding to the total number.

Further, the sound source state determination section 70 has the level P of each channel detected by the all-band level detection section 60.
If (S1) to P (S3) are also input and all of the levels are equal to or less than the preset reference value ThR, it is determined that there is no sound source in any zone. Based on the determination result by the sound source state determination unit 70, a control signal is generated, and the signal suppression unit 90 suppresses the sound signals A and B divided by the sound source separation unit 80. That is, the control signal SAi
Suppresses (attenuates or deletes) the sound signal SA, suppresses the sound signal SB with the control signal SBi, and outputs the control signal SA.
Bi suppresses both acoustic signals SA and SB. For example, normally closed switches 9A and 9B are provided in the signal suppression unit 90,
The output terminals t _A and t _{B of the} sound source separation unit 80 are normally closed switches 9
A, 9B are connected to the output terminals t _A ′, t _B ′, the switch 9A is opened by the control signal SAi, the switch 9B is opened by the control signal SBi, and both the switches 9A, 9B are controlled by the control signal SABi. It is opened.
As a matter of course, the same signal is used for the signal of the frame to be separated by the sound source separation unit 80 and the signal of the frame for obtaining the control signal used for suppression by the signal suppression unit 90. The generation of the suppression (control) signals SAi, SBi, and SABi will be described in an easily understandable manner.

Now, when sound sources A and B are located as shown in FIG. 15, microphones M1 to M3 are arranged as shown in the figure, zones Z1 to Z6 are determined, and sound sources A and B are separated. It should be located in each of the zones Z3 and Z4.
At this time, the distances SA1, SA2, and SA3 of the sound source A to the microphones M1 to M3 satisfy SA2 <SA3 <SA1. The distances SB1, SB2, and SB3 of the sound source B from the microphones M1 to M3 are represented by SB3 <SB2 <S.
B1.

The detection signal P (S
When all of 1) to P (S3) are smaller than the reference value ThR, it is considered that the sound sources A and B do not generate sound, for example, speak, and the control signal SABi suppresses both sound signals SA and SB. At this time, the output acoustic signals SA and SB are silent signals (101 and 102 in FIG. 16). When only the sound source A is sounding, the frequency components in all the bands of the sound signal reach the microphone M2 at the highest sound pressure level (power), so that the total number of channels χ2 of the channels of the microphone M2 is The most.

When only the sound source B is sounding,
Since the frequency components of all the bands of the acoustic signal reach the microphone M3 at the highest sound pressure level, the total number of bands χ3 of the channels of the microphone M3 is the largest. Further, when both of the sound sources A and B are sounding, the microphones M2 and M3 compete with each other for the number of bands in which the sound signal reaches at the highest sound pressure level.

Therefore, according to the above-mentioned reference value ThP, if the total number of bands in which the acoustic signal reaches the microphone with the largest sound pressure level exceeds the reference value ThP, it is determined that the sound source exists in the zone controlled by the microphone. By making the determination, the sound source zone that is sounding can be detected. In the above example, when only the sound source A is sounding, only # 2 exceeds the reference value ThP, and it is detected that the sounding sound source exists in the zone Z3 controlled by the microphone M2. The audio signal SB is suppressed by the control signal SBi, and only the audio signal SA is output (103 and 104 in FIG. 16).

Further, if the sound sources A and B are both sounding,
When both # 2 and # 3 exceed the reference value ThP, for example, priority can be given to the sound source A, and it can be processed that only the sound source A is sounding. The processing procedure of FIG. If both # 2 and # 3 do not reach the reference value ThP, it is determined that both sound sources A and B are sounding as long as the levels P (S1) to P (S3) exceed the reference value ThR. , And does not output any of the control signals SAi, SBi, and SABi.
B is not suppressed (107 in FIG. 16).

As described above, the sound source signals SA and SB separated by the sound source separation unit 80 correspond to the sound source determined not to be sounding by the sound source state determination unit 70, and are output by the signal suppression unit 90. It is suppressed, and unnecessary sound is suppressed. The generation of such a control signal can also be detected using the difference in arrival time between bands. That is, in FIG. 14, the inter-band level difference detection unit 51 detects arrival time difference signals An (S1f1) to An (S1fn) instead of the level difference, and similarly arrives at the time difference signals An (S2f1) to An.
In the process of detecting (S2fn) and An (S3f1) to An (S3fn) and obtaining the arrival time difference signal, for example, the phase (or group delay) of the signal in each band is calculated by Fourier transform, and the same band is calculated. fi signal S1 (f
i), S2 (fi), S3 (fi) (i = 1, 2,...,
By comparing the phases n) with each other, a signal corresponding to the arrival time difference of the same sound source signal can be obtained. In this case as well, the division by the band dividing section 40 is performed so small that only one sound source signal component exists in one band.

The method of expressing the arrival time difference is as follows. For example, if the arrival time difference with respect to one of the microphones M1 to M3 is set to 0, the arrival time difference with respect to the other microphones is set to the reference microphone. On the other hand, since it is possible to determine whether the vehicle has arrived fast or late, it can be represented by a numerical value with a positive or negative polarity. In this case, assuming that the reference microphone is M1, for example, arrival time difference signals An (S1f1) to An (S1
fn) are all 0.

In the sound source state determination unit 70, the arrival time difference signals An (S1f1) to An (S1fn), An (S2f)
1) to An (S2fn), An (S3f1) to An (S
3fn) are compared with each other for signals in the same band. As a result, for each of the bands f1 to fn, the channel in which the signal reaches the fastest can be determined. Therefore, the total number of bands determined to reach the fastest signal for each channel is calculated and compared between channels. As a result, the microphone of a channel having a larger value of the total number of bands can be regarded as being closer to the sound source. Then, for a certain channel, when the total number of bands exceeds a preset reference value ThP, it is determined that a sound source exists in a zone controlled by the microphone of the channel.

Assume that microphones M1 to M3 are arranged for sound sources A and B as shown in FIG. The total number of bands for the channel of the microphone M1 is $ 1, and the total number of bands for each channel of the microphones M2 and M3 is $ 2 and $ 3, respectively. In this case, the procedure may be the same as the procedure shown in FIG. That is, first, the detection signals P (S1) to P (P1)
When all of (S3) is smaller than the reference value ThR (1
01), it is considered that the sound sources A and B are not sounding, and a control signal SABi is generated (102).
Suppress B. At this time, the output signals SA 'and SB' are silent signals.

When only the sound source A is sounding, the frequency components of all the bands of the sound source signal are output from the microphone M2.
, The total number of bands χ2 of the channel of the microphone M2 becomes the largest. Further, when only the sound source B is sounding, the frequency components of all the bands of the sound source signal reach the microphone M3 fastest, so that the total number of bands of the channel of the microphone M3χ3
Is the most.

Further, when both the sound sources A and B are sounding, the number of bands in which the sound source signal reaches the fastest is opposed by the microphones M2 and M3. Therefore, when the total number of bands in which the sound source signal reaches the microphone at the earliest exceeds the set value ThP, the sound source exists in the zone controlled by the microphone, and the sound source emits a sound. It is determined that there is.

In the above example, when only the sound source A is sounding, only # 2 exceeds the reference value ThP (1 in FIG. 16).
03) Since it is detected that the sound source generating the sound exists in the zone Z3 controlled by the microphone M2,
A control signal SBi is generated (104), the acoustic signal SB is suppressed, and only the signal SA is output. Further, when only the sound source B is sounding, only # 3 exceeds the reference value ThP (105), and it is detected that the sound source emitting the sound is in the zone Z4 controlled by the microphone M3. , A control signal SAi is generated (106), the signal SA is suppressed, and only the signal SB is output.

In this example, ThP is set to, for example, about n / 3, and both the sound sources A and B are sounding, and both # 2 and # 3 may exceed the reference value ThP. In this case, FIG.
As shown in the processing procedure 3, one of the sound sources, A in this example, may be prioritized, and only the separated signal may be output to the sound source A. When both # 2 and # 3 do not reach the reference value ThP, the levels P (S1) to P (S3) are changed to the reference value TP.
As long as it exceeds hR, it is determined that both sound sources A and B are sounding, and the control signals SAi, SBi and SABi are not output (107 in FIG. 16).
No suppression is performed on A and SB.

FIG. 17 is a functional block diagram of an example in which the method of suppressing or silencing a synthetic sound signal that is not sounding is applied to a wraparound suppression sound pickup apparatus, and corresponds to FIGS. 2, 12, and 14. The same reference numerals are given to the portions to be performed. That is, in this case, each channel signal from the microphones 1 and 2 is divided into a plurality of bands by the band division unit 4, and the audio signal selection unit 602L, the band-by-channel time difference / level difference detection unit 5, the band-by-band level / time difference It is supplied to the detection unit 50. The outputs of the two microphones 1 and 2 are also supplied to an inter-channel time difference / level difference detection unit 3, and the inter-channel time difference or level difference is supplied to a band-specific inter-channel time difference / level difference detection unit 5 and an audio signal determination unit 601. The output levels of the microphones 1 and 2 are supplied to the sound source state determination unit 70.

The output of the inter-channel time difference / level difference detection unit 5 for each band is supplied to the audio signal judgment unit 601, and as described above, which sound source component is judged for each band.
Based on the determination result, the audio signal selection unit 602L selects only the audio signal component of the specific sound source, in this example, only the component of the voice of one speaker, and supplies it to the sound source signal synthesis unit 7. On the other hand, the level / arrival time difference of each band is detected by the band-specific level / time difference detection unit 50, and these detection outputs are detected by the sound source state determination unit 70 by detecting the sound source that is sounding or not as described above. Then, the synthesized sound source signal that is not sounding is suppressed by the signal suppression unit 90.

The technique of suppressing a synthetic sound source signal that is not sounding can be applied to the wraparound suppression sound pickup apparatus shown in FIGS. The band division unit 4 in FIG.
Each band division unit 40 in FIG. 14 and band division unit 2 in FIG.
33, the division of each frequency band in the band division unit 241 in FIG. 13 does not necessarily have to be the same. These division numbers may be different from each other depending on the required accuracy. The band division unit 4, the band division unit 233 in FIG. 12, and the band division unit 40 in FIG. 14 in the case of using the inter-band level difference in FIG. The frequency band may be determined first, and then divided into a plurality of frequency bands.

[0074]

As described above, according to the present invention,
The output signals of the plurality of microphones are divided into a plurality of bands that are sufficiently narrow, the parameter values of the acoustic signal in each band are detected, the difference between the same bands is detected, and the parameter value difference is determined by a threshold. Compared to the value, the speaker's voice signal can be correctly separated from other acoustic signals, and howling can be sufficiently suppressed with a relatively simple configuration. Moreover, there is little deterioration of sound.

The received signal is divided into a plurality of bands that are sufficiently narrow so that there is a band whose level (power) in the band is sufficiently negligible, and only the band in which the signal can be ignored is separated and extracted. The component of the band to be synthesized and transmitted is removed from the received signal subjected to voice synthesis or band division, and the removed divided band received signal is subjected to voice synthesis to obtain an electroacoustic converter. By supplying to the hopper, howling can be suppressed more reliably.

[Brief description of the drawings]

FIG. 1 is a block diagram showing a main configuration of a device of the present invention.

FIG. 2 is a block diagram showing a functional configuration of an embodiment of a sound source separation unit used in the present invention.

FIG. 3 is a flowchart showing a processing procedure of an embodiment of a sound source separation method used in the present invention.

FIG. 4 is a flowchart showing an example of a processing procedure for obtaining time differences Δτ ₁ and Δτ ₂ between channels in FIG. 3;

FIGS. 5A and 5B are diagrams illustrating examples of spectra of two sound source signals, respectively.

FIG. 6 is a flowchart showing a processing procedure of an embodiment in which sound source separation is performed using a level difference between channels in a sound source separation method.

FIG. 7 is a flowchart showing a part of a processing procedure of an embodiment using an inter-channel level difference and an inter-channel arrival time difference in a sound source separation method.

FIG. 8 is a flowchart showing a continuation of step S08 in FIG. 7;

FIG. 9 is a flowchart showing a continuation of step S09 in FIG. 7;

FIG. 10 is a flowchart showing a continuation of step S10 in FIG. 7 and steps S20 and S30 in FIGS. 7 and 8;

FIG. 11 is a block diagram showing a functional configuration of an embodiment for separating sound source signals having different frequency bands.

FIG. 12 is a block diagram showing a functional configuration of an embodiment of the sneak noise suppression type sound collecting device according to the present invention.

FIG. 13 is a block diagram showing a part of the functional configuration of another embodiment.

FIG. 14 is a block diagram showing a functional configuration of an embodiment of a sound source separation unit to which a structure for suppressing an unnecessary sound source signal by using a level difference is added.

FIG. 15 is a diagram showing an example of the arrangement of three microphones, their zones, and two sound sources.

FIG. 16 is a flowchart showing an example of a procedure for detecting a sound source zone and generating a suppression control signal when only one sound source is sounding;

FIG. 17 is a block diagram showing a functional configuration of still another embodiment of the present invention.

 ──────────────────────────────────────────────────の Continuing on the front page (72) Inventor Hiroyuki Matsui Nippon Telegraph and Telephone Corporation, 3-19-2 Nishi Shinjuku, Shinjuku-ku, Tokyo

Claims (35)

[Claims] 1. A microphone which picks up a voice signal from a sound space in which a sound signal obtained by converting a reception signal from the ground by an electro-acoustic conversion means and a voice signal from a speaker exist, and In the method of collecting the sound signal by suppressing the wraparound of the acoustic signal of the sound pickup device to be transmitted to the plurality of microphones, a plurality of microphones are provided apart from each other as the microphone, and each output channel signal of the microphone is provided by a plurality. Band-specific parameters for detecting a parameter value of an acoustic signal arriving at the microphone, which varies according to the positions of the plurality of microphones, for each band of each of the band-divided channel signals. Value detection process, and a parameter for detecting a difference between channels of the detected parameter value for each same band. A value difference detection step, and using these detected parameter value differences, based on a preset threshold value, an audio signal selection for selecting the audio signal component from the band-divided channel signal in the unit of the band. A voice synthesis process for synthesizing voice signal components of these selected bands into a voice signal; and transmitting the synthesized voice signal to the ground. 2. The method according to claim 1, wherein the received signal is divided into a plurality of frequency bands, and a level detecting step of detecting the level of each band of the band-divided received signal; For each band of the received signal, if the level is equal to or less than a predetermined value, a transmittable band determining step of determining a transmittable band, and the transmittable band signal selected in the audio signal selecting step is determined to be transmittable. And a transmission selectable step of selecting only the selected band and sending the selected band to the speech synthesis step. 3. The method according to claim 2, wherein the selection control in the transmission enabled selection step is performed with a delay corresponding to a propagation time of an acoustic signal between the electroacoustic transducer and the microphone. A wraparound noise suppression type sound collection method having a delay process. 4. The method according to claim 1, wherein the received signal is divided into a plurality of frequency bands, a second band dividing step, and the band division corresponding to the band selected in the audio signal selecting step. And a re-synthesizing step of combining the band components of the remaining received signal from which the components have been removed with a signal in the time domain and supplying the signal to the electroacoustic conversion means. Sound suppression type sound collection method. 5. The method according to claim 1, wherein each of the channel signals is divided into a plurality of bands, and a position of the plurality of microphones is set for each band of the divided signals. A second band-based parameter value detecting step of detecting a parameter value of an acoustic signal reaching a microphone which is changed due to the cause, and uttering an utterance based on a result of comparing the detected band-based parameter values in the same band between channels. A sound source state determination step of detecting a speaker who has not performed speech, and a detection signal for detecting a speaker who has not spoken obtained in the sound source state determination step. A sneaking sound suppressing type sound collecting method, comprising: a speaker who is not speaking and a signal suppressing process of suppressing a synthesized signal corresponding to the speaker. 6. The method according to claim 1, wherein the parameter value in the step of detecting a parameter value difference between channels for each band is determined until a sound signal from a sound source reaches each of the microphones. Wherein the inter-channel parameter value difference is a time difference between the microphones in the time required to reach each microphone. 7. The method according to claim 1, wherein the parameter value in the step of detecting a parameter value difference between channels for each band is a signal level when an acoustic signal from a sound source reaches each of the microphones. Wherein the inter-channel parameter value difference is a band-to-channel level difference which is a level difference between corresponding bands of the divided output channel signals. 8. The method according to claim 1, wherein the parameter value in the band-to-channel parameter value difference detecting process is a time required for an acoustic signal from a sound source to reach the microphone. It is the signal level when the sound signal arrives, and the band-to-channel parameter value difference is the band-to-channel time difference, and the band-to-channel level difference is obtained. Dividing each output channel signal into three frequency regions of a low band, a middle band, and a high band, and performing a region dividing process. The audio signal selecting process includes: The above-mentioned selection is performed using the time difference between different channels, and for the divided middle frequency band, the level difference between channels for each band and the channel difference for each band are described. The above-mentioned selection is performed by using the time difference between channels, and for the divided high frequency band, the above-described selection is performed by using the above-mentioned channel-by-band level difference between bands. Sound collection method. 9. The wraparound sound according to claim 1, further comprising a step of silencing a band having a level equal to or less than a predetermined value with respect to the band-divided channel signal. Suppressed sound collection method. 10. The method according to claim 1, further comprising a step of excluding, when the parameter value difference between the bands is larger than a reference value, the selection process. Wraparound sound suppression type sound collection method. 11. The wraparound sound according to claim 1, further comprising a step of removing a delayed wraparound signal from the synthesized speech signal synthesized by the speech synthesis step. Suppressed sound collection method. 12. The method according to any one of claims 1 to 8, wherein the band division in the process of detecting the band-specific parameters includes a band as narrow as possible so that only a signal component of one sound source is included in each band. A wraparound noise suppression type sound collecting method. 13. The method according to claim 1, wherein the band division in the level detection step is a band narrow enough that each band mainly includes only a signal component of one sound source. A wraparound noise suppression type sound collection method characterized by the following. 14. The method according to any one of claims 1 to 8, 12 and 13, wherein the band division in the second band-specific parameter value detection process includes only one sound source signal component in each band. A sneak noise suppression type sound collection method characterized by a narrow band. 15. A microphone that picks up the audio signal from a sound space in which a sound signal obtained by converting a reception signal from the ground by an electroacoustic conversion unit and a voice signal from a speaker exist, and A plurality of microphones provided apart from each other as the microphone, and each output channel signal of these microphones is divided into a plurality of frequency bands, and each of the band-divided channel signals is For each band, a parameter value detecting means for each band which detects a parameter value of an acoustic signal reaching the microphone, which changes due to the positions of the plurality of microphones, and a parameter value detecting means for each of the same bands. Parameter value difference detecting means for detecting a difference between channels, and using the detected parameter value differences, An audio signal selecting means for selecting the audio signal component from the band-divided channel signal based on a preset threshold value in band units; and converting the audio signal component of the selected band into an audio signal. A wraparound sound suppression type sound collection device comprising: a voice synthesis unit for synthesizing; and a unit for transmitting the synthesized voice signal to the ground. 16. The apparatus according to claim 15, wherein the received signal is divided into a plurality of frequency bands,
Level detecting means for detecting the level of each band of the divided received signal; and transmittable band determining means for judging the transmittable band if the level of each band of the band-divided received signal is equal to or less than a predetermined level. And a transmission selectable means for selecting only a band determined to be transmittable in the band signal selected by the audio signal selection means and transmitting the selected band to the voice synthesis process. 17. The apparatus according to claim 16, wherein the selection control by said transmittable selection means is performed with a delay corresponding to a propagation time of an acoustic signal between said electroacoustic conversion means and said microphone. A wraparound sound suppressing type sound collecting device having a delay means for delaying. 18. The apparatus according to claim 1, wherein said received signal is divided into a plurality of frequency bands by a second band dividing means, and said band division corresponding to the band selected by said sound source signal selecting means. Frequency component removing means for removing the received signal, and re-synthesizing means for combining the band component of the remaining received signal from which the component has been removed into a time-domain signal and supplying the signal to the electroacoustic converting means. Shape sound pickup device. 19. The apparatus according to claim 15, wherein the parameter value in the band-specific inter-channel parameter value difference detecting means is a time until an acoustic signal from a sound source reaches each of the microphones. And wherein the inter-channel parameter value difference is a band-to-channel inter-channel time difference which is a difference between microphones in a time required to reach each microphone. 20. The apparatus according to claim 15, wherein said parameter value in said band-by-band parameter value difference detecting means is a signal level when an acoustic signal from a sound source reaches each of said microphones. Wherein the inter-channel parameter value difference is a band-to-channel level difference which is a level difference between corresponding bands of the divided output channel signals. 21. The apparatus according to claim 15, wherein the parameter value in the band-specific inter-channel parameter value difference detecting means is a time required for an acoustic signal from a sound source to reach the microphone. The signal level when the sound signal arrives, the inter-channel parameter value difference by band is the inter-channel time difference by band, and the inter-channel level difference by band, and the division is performed based on the inter-channel time difference. And each of the output channel signals is divided into three frequency regions of a low band, a middle band, and a high band. The audio signal selecting unit includes: The above-mentioned selection is performed by using the time difference between the channels for each band, and for the divided middle frequency band, the level difference between the channels for each band and the channel difference for each band are obtained. Means for making the selection using the time difference between channels, and for the divided high frequency band, means for making the selection by using the level difference between the channels for each band. Sound suppression type sound collection device. 22. The apparatus according to claim 15, wherein each of the channel signals is divided into a plurality of frequency bands, and the position of the plurality of microphones is determined for each of the divided signal bands. A second band parameter value detecting means for detecting a parameter value of an acoustic signal reaching a microphone that changes due to the above, based on a result of comparing the detected band parameter values between channels for the same band. Sound source state determination means for detecting a speaker who is not speaking; and a detection signal for detecting a speaker who is not speaking obtained by the sound source state determination means. And a signal suppression unit for suppressing the synthesized signal corresponding to the speaker who is not speaking. 23. The apparatus according to claim 15, further comprising a threshold value setting unit that sets the threshold value in the audio signal selection unit. 24. The apparatus according to claim 15, wherein, in the audio signal selecting means, a signal for which the parameter value difference between channels for each band is larger than a reference value is excluded from a determination target. A wraparound noise suppression type sound collection device having a means for setting a value. 25. The apparatus according to claim 15, wherein said audio signal selecting means includes reference value setting means for setting a criterion for determining a band component having a level equal to or less than a predetermined value as silence. Sound suppression type sound collection device. 26. The apparatus according to claim 15, further comprising removing means for removing a delayed sneak signal from the synthesized signal from the sound source signal synthesizing means. 27. A microphone which picks up the sound signal from a sound space in which a sound signal obtained by converting a received signal from the ground by an electro-acoustic conversion means and a sound signal from a speaker exist, and A recording medium that records a program that suppresses the wraparound of the acoustic signal in the sound collection device that transmits the sound signal and collects and transmits the sound signal, wherein the program is configured to convert each output channel signal of each microphone into a plurality of microphones. The frequency band is divided into frequency bands, and for each band of each of the divided output channel signals, a difference between parameter values of an acoustic signal reaching the microphones, which varies due to the positions of the plurality of microphones, is classified by band. A band-by-band parameter value difference detection process for detecting as a channel-to-channel parameter value difference; An audio signal selection step of selecting the audio signal component in the band unit from the band-divided output channel signal based on a threshold value set in advance using the difference, and an audio signal component selected in the audio signal selection step. A computer-readable recording medium having a voice synthesizing step of synthesizing voice signal components of a plurality of bands as a voice signal, and a step of transmitting the synthesized voice signal to the ground. 28. The recording medium according to claim 27, wherein the program divides the reception signal into a plurality of frequency bands and detects a level of the reception signal in each of the divided bands. A transmittable band determining step of determining a transmittable band if the detection level is equal to or less than a predetermined value, for each band of the band-divided received signal, and in the band signal selected in the audio signal selecting step, A transmission possible selecting step of selecting only the band determined to be transmittable and transmitting the selected band to the sound source synthesizing step. 29. The recording medium according to claim 28, wherein the program performs selection control in the transmittable selection step in accordance with a propagation time of an acoustic signal between the electroacoustic conversion unit and the microphone. A recording medium having a delay process to be performed with a delay. 30. The recording medium according to claim 27, wherein the program corresponds to a second band division step of dividing the reception signal into a plurality of frequency bands, and a band selected in the audio signal selection step. A frequency component removing step of removing the band-divided received signal, and combining the remaining band component of the band-divided received signal from which the component has been removed with a time-domain signal and supplying the time-domain signal to the electroacoustic conversion means. A recording medium having a re-synthesis process. 31. The recording medium according to claim 27, wherein the parameter value in the band-by-band parameter value difference detection process is a time required for an acoustic signal from a sound source to reach each of the microphones. Wherein the difference between band-specific parameter values is a time difference between channels, which is a difference between microphones in a time required to reach each microphone. 32. The recording medium according to claim 27, wherein the parameter value in the band-by-band parameter value difference detection process is a signal when an acoustic signal from a sound source reaches each of the microphones. The recording medium according to claim 1, wherein the parameter value difference between bands is a level difference between channels, which is a level difference between corresponding bands of the divided output channel signals. 33. The recording medium according to claim 27, wherein the parameter value is a time until an acoustic signal from a sound source reaches the microphone and a signal level when the acoustic signal reaches the microphone. Wherein the inter-channel parameter value difference by band is a time difference between channels by band and a level difference between channels by band.Based on the time difference between channels, each of the divided output channel signals is And a region dividing step of dividing the frequency band into three frequency regions of a middle band and a high band. The audio signal selecting step is characterized in that, for the divided low band frequency band, using the time difference between channels for each band. Selection, and for the divided middle frequency band, the selection is performed using the inter-channel level difference between the bands and the inter-channel time difference between the bands. The frequency band of the divided high frequency band, using the level difference between the per-band channel, a recording medium which is a process of performing the selection. 34. The recording medium according to claim 27, wherein the program divides each channel signal into a plurality of frequency bands and, for each band of the divided signals, A second band parameter value detection process of detecting a parameter value of an acoustic signal reaching the microphone that changes due to the microphone position, and a result of comparing these detected band parameter values between channels in the same band. A sound signal synthesized in the above-described speech synthesis process by a sound source state determination process for detecting a speaker who is not speaking based on the sound source state detection process and a detection signal for detecting a speaker who is not uttering obtained in the sound source state determination process. And a signal suppressing step of suppressing a synthesized signal corresponding to the speaker who is not speaking. 35. The recording medium according to claim 27, wherein the program includes a step of removing a delayed wraparound signal from the synthesized speech signal synthesized by the speech synthesis step. Characteristic recording medium.