JP3159229B2 - Microphone device - Google Patents

Description

DETAILED DESCRIPTION OF THE INVENTION

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a so-called super-directional microphone device using an adaptive noise canceller.

[0002]

2. Description of the Related Art For example, in a camera-integrated VTR, sounds around a subject are simultaneously recorded while photographing the subject. When picking up this sound,
Generally, it is considered that only sound from the direction of the subject is picked up. That is, for example, a microphone device having a directional characteristic of collecting only a sound from the front of the camera is used.

[0003] As an example of this kind of microphone device,
For example, a so-called gun microphone is known. It has a pipe section extending in front of the diaphragm. A large number of through holes are provided in the side wall of the pipe portion, and have directivity having high sensitivity to sound from the front in the center line direction of the pipe portion (the direction opposite to the diaphragm). It is configured as follows.

[0004]

However, this microphone has a drawback that it requires a long pipe section and is large in size. In addition, it is unidirectional with high sensitivity only in front of the microphone, and only fixed directivity can be obtained. For this reason, it is difficult to cope with a case where it is desired to collect not only the sound from the desired sound arrival direction but also the sound from the side around the camera, for example, and there is no flexibility in the direction of directivity.

Accordingly, the applicant has filed a Japanese Patent Application No. Hei 4-14320.
No. 9 proposes a microphone device that can be configured to have a small size and super directional characteristics by applying an adaptive noise canceller.

FIG. 4 shows a basic configuration of an adaptive noise canceller. First, the adaptive noise canceller will be described.

In FIG. 1, reference numeral 1 denotes a main input terminal, and 2 denotes a reference input terminal. A main input signal input through the main input terminal 1 is supplied to a synthesizing circuit 4 via a delay circuit 3. Further, the reference input signal input through the reference input terminal 2 is supplied to the synthesizing circuit 4 via the adaptive filter circuit 5, and is subtracted from the signal from the delay circuit 3. The output of the synthesizing circuit 4 is fed back to the adaptive filter circuit 5 and is output to an output terminal 6.

In this adaptive noise canceller, a signal obtained by adding a desired signal s and an uncorrelated noise signal n0 is input as a main input signal. On the other hand, a noise signal n1 is input as a reference input signal. The noise signal n1 of the reference input is uncorrelated with the desired signal s, but is correlated with the noise signal n0.

The adaptive filter circuit 5 filters the reference input noise signal n1 and outputs a signal y approximating the noise signal n0. In this case, in the adaptive filter circuit 5, the combining circuit 4
Is updated so as to minimize the power of the subtraction output (residual output) e of the reference input noise signal n1.

As the output signal y of the adaptive filter circuit 5, a signal having the same phase and the same amplitude as the noise signal n0 can be obtained. The delay circuit 3 compensates for a time delay required for the arithmetic processing in the adaptive filter circuit 5, a propagation time in the adaptive filter, and the like, and adjusts the time with the signal to be subjected to the subtraction processing.

The principle of the adaptive noise canceller will be described below.

Assuming that the desired signal s, the noise n0, the noise n1, and the output signal y are statistically stationary and the average value is zero, the residual output e becomes e = s + n0-y. The expected value of the squared value is E [e ² ] = E [s ² ] + E [(n 0 −y) ² ] + 2E [s (n 0) because the desired signal s is uncorrelated with the noise n 0 and the output y. -y)] = a ^{E [s 2] + E [} (n0 -y) 2]. Assuming that the adaptive filter circuit 5 converges,
The adaptive filter circuit 5 updates the adaptive filter coefficients so that E [e ² ] is minimized. At this time, since the E [s ^2] is not affected, and ^{Emin [e 2] = E [} s 2] + Emin [(n0 -y) 2].

That is, by minimizing E [e ² ], E [(n 0 −y) ² ] is minimized, and the output y of the adaptive filter circuit 5 becomes the best estimate of the noise signal n 0. . The expected value of the output from the synthesizing circuit 4 is only the desired signal s. That is, adjusting the adaptive filter circuit 5 to minimize the total output power is equivalent to the subtraction output e
Is equivalent to a least-squares estimate of the desired speech signal s.

The output e generally has some noise remaining in the signal s. However, since the output noise is given by (n0-y), it is not possible to minimize E [(n0-y) ² ]. , Which is equivalent to maximizing the signal-to-noise ratio of the output.

In some cases, the synthesizing circuit 4 serves as sound synthesizing means. That is, the adaptive filter circuit 5 forms a noise-cancelling audio signal -y having a phase opposite to that of noise and equal amplitude, supplies the signal to a speaker or the like, and acoustically adds the noise to the main audio to reduce noise. I do. The residual e in this case is picked up by the residual detection microphone.

The adaptive filter circuit 5 can be realized by either an analog signal processing circuit or a digital signal processing circuit.
A digital processing circuit using an SP (Digital Signal Processor) is used. FIG. 6 shows an example of the configuration of the adaptive filter circuit 5 in the case of a digital processing circuit.

The adaptive filter circuit 5 in this example comprises an FIR filter type adaptive linear combiner 100 and a filter coefficient updating operation means 110. The adaptive filter circuit 5 can be configured as software by a DSP equipped with a microcomputer. In this example, the algorithm for updating the filter coefficients will be described assuming that the LMS (Least Mean Squares) method that is frequently used because of its small amount of calculation and practicality is used.

The LMS method will be described with reference to FIG. As shown in FIG. 6, an FIR filter type adaptive linear combiner 100 is used in this case. The adaptive linear combiner 100 includes a plurality of delay circuits DL1, DL2, each having a delay time Z ⁻¹ of a unit sampling time.
.., DLm (m is a positive integer), weighting circuits MX0, which multiply input noise n1 and output signals of delay circuits DL1, DL2,.
MX1, MX2,..., MXm and weighting circuits MX0 to MX
An addition circuit 101 for adding the outputs of m is provided. The output of the adding circuit 101 is the signal y described in FIG.

The weighting coefficients supplied to the weighting circuits MX0 to MXm are formed in the filter coefficient calculation circuit 110 by the LMS algorithm based on the residual signal e from the synthesis circuit 4 and the reference input n1. The algorithm executed by the filter coefficient operation circuit 110 is as follows.

Now, the input vector X _k at time k is
As shown in FIG. 6, X _k = [x _0k x _1k x _2k ... X _mk ] ^T , the output is y _k , and the weighting factor is w _jk (j = 0, 1, 2,... M). Then, the relationship between the input and output is as shown in the following equation 1.

[0021]

(Equation 1) Becomes

Then, the weight vector W _{k at} time _k
Is defined as W _k = [w _{0 k} w _{1 k} w _{2 k} ... W _mk ] ^T , the input / output relationship is given by y _k = X _k ^T · W _k (1) Here, assuming that the desired response is d _k , the residual error e _k is represented by e _k = d _k −y _k = d _k −X _k ^T · W _k (2)

[0023] In the LMS method, the update of the weight _{vector, W k + 1 = W k} + is sequentially carried out by _{2μ · e k · X k ...} (3) consisting of formula (3). The initial value of the weighting coefficient is set to a constant value or a random value. Here, μ is a gain factor (step gain) that determines the speed and stability of adaptation.

The microphone device proposed above is configured using the adaptive noise canceller as described above.
That is, as shown in FIG. 4, a first microphone 7 for main input sound pickup and a second microphone 8 for reference input sound pickup are provided, and an output signal of the first microphone 7 is supplied to the main input terminal 1. And the output signal of the second microphone 8 is input to the reference input terminal 2.

In this case, when the desired sound arrival direction is AR, the main input microphone 7 is
An omnidirectional microphone as shown in Fig. 1 is used. Alternatively, a unidirectional microphone is arranged with its main directional axis oriented in the direction from which the desired sound comes.

Further, the reference input microphone 8 is arranged as shown in FIG.
As shown in (1), microphones having unidirectionality are used by arranging them so that they have sensitivity in the voice input direction to be eliminated and that the desired voice arrival direction AR is the null direction of the directivity. The direction of arrival of the voice to be eliminated is 90
In the case of the degree direction, the bidirectional microphones may be arranged and used so that the desired sound arrival direction AR becomes the directional null direction.

According to this configuration, the reference input signal hardly contains the desired voice signal. Therefore, the reference input signal is a signal that is uncorrelated with the desired voice, and an unnecessary signal (noise) in the main input is output. Component). Therefore, if the second microphone 8 for collecting the reference input is configured to have a predetermined sensitivity in the arrival direction of the unnecessary signal to be reduced, the unnecessary signal component included in the main input is adaptively canceled. Then, only the desired audio signal is obtained at the output terminal.

As described above, using the adaptive noise canceller, it is possible to realize a super-directional microphone device having sharp directivity in the direction of the minimum sensitivity of the unidirectional microphone 8 for reference input. it can.

However, as a practical problem, it is difficult to make the directional characteristics of the microphone for reference input sound collection such that the desired sound signal is not completely picked up (the sensitivity is set to zero). For this reason, it is inevitable that the desired audio signal is mixed into the reference input signal at a certain level.

This state deviates from the precondition of the adaptive processing that both input signals of the desired voice and the reference input voice are uncorrelated. In particular, when the level of the unnecessary signal (noise) of the reference input is considerably low. Is that in the normal adaptive processing as described above, the desired audio signal itself is a target of reduction,
There is a problem that a desired audio signal is deteriorated.

In view of the above, it is an object of the present invention to provide a microphone device using an adaptive noise canceller, which can reduce deterioration of a desired voice signal.

[0032]

In order to solve the above-mentioned problems, a microphone device according to the present invention comprises first and second microphone devices.
Microphone, filter means to which the output signal of the first microphone is supplied, subtraction means for subtracting the output signal of the filter means from the output signal of the second microphone, and a noise component included in the main input signal And an adaptive noise canceller that adaptively reduces the noise based on the reference input signal.
When a signal for identification is input from the direction of maximum sensitivity of the directivity to be created and collected by the first and second microphones, the characteristic of the result of identification is such that the output of the subtraction means becomes zero, An output signal of the first microphone is supplied to the adaptive noise canceller as the main input signal, and an output signal of the subtraction means is supplied to the adaptive noise canceller as a reference input.

[0033]

In the present invention having the above-described structure, the characteristic of the output signal of the subtracting means is such that the signal has the minimum sensitivity (zero) with respect to the signal from the directional direction to be created, ie, from the desired voice direction. Becomes That is, a signal equivalent to that obtained by the microphone whose sensitivity in the desired voice direction becomes zero is obtained from the subtraction means. Therefore, it is possible to prevent the desired sound from being mixed into the output of the subtraction means serving as the reference input of the adaptive noise canceller.

[0034]

DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS One embodiment of a microphone device according to the present invention will be described below with reference to FIG. This example is an example where the adaptive filter circuit of the adaptive noise canceller has a digital configuration.

In FIG. 1, reference numerals 11 and 12 denote first and second microphones. In this example, both microphones are omnidirectional microphones. Reference numeral 10 denotes an adaptive noise canceller, and the adaptive filter circuit 5 includes, as described above,
Adaptive linear combiner 100 and filter coefficient operation circuit 11
0.

The sound is picked up by the first microphone 11,
An audio signal obtained by being converted into an electric signal is converted into a digital signal in an A / D converter 13 and used as a main input signal of a main input terminal 1 of the adaptive noise canceller 10 as a main input signal.
Supplied to

On the other hand, the sound signal collected by the second microphone 12 and converted into an electric signal is
The signal is converted into a digital signal by the D converter 14 and supplied to the subtraction circuit 15. This subtraction circuit 15 has A /
A digital signal output from the first microphone 11 from the D converter 13 is supplied through an FIR filter circuit 21 having the same configuration as that of the adaptive linear combiner 100.

The subtraction circuit 15 converts the digital signal from the A / D converter 14 into an FIR filter circuit 2.
1 is subtracted, and the subtracted output is supplied to the reference input terminal 2 of the adaptive noise canceller 10 as a reference input. The output signal of adaptive noise canceller 10 is D /
The analog signal is returned to the analog signal by the A converter 16 and is output to the output terminal 17.

The filter coefficients of the FIR filter circuit 21 are supplied from a filter coefficient calculation supply circuit 22. The filter coefficient operation supply circuit 22 can be configured by a DSP, similarly to the filter coefficient operation circuit 110 of the adaptive noise canceller 10. In addition, the adaptive noise canceller 10
Can also be used as the filter coefficient calculation circuit 110 of FIG.

The value of the filter coefficient is obtained as described later. The first and second microphones 11 and 12 receive the identification signal from the directivity direction to be created.
Is the value of the filter coefficient as a result of identification so that the output of the subtraction circuit 15 becomes zero when sound is collected.

A digital signal of the output of the first microphone 11 from the A / D converter 13 is supplied to the filter coefficient calculation supply circuit 22 via the switch SW1, and the output of the subtraction circuit 15 is supplied via the switch SW2. A signal is provided. The switches SW1 and SW2 are turned on when obtaining a filter coefficient to be supplied to the FIR filter circuit 21. After the filter coefficients are obtained, the switches SW1 and SW2 are turned off, and the obtained filter coefficients are applied to the FIR filter circuit 21.
Is fixed as the filter coefficient value.

The operation of the filter coefficient calculation supply circuit 22 and the method of determining the filter coefficient of the FIR filter circuit 21 will be described. First, the switches SW1 and SW2 are turned on. Then, the identification signal is input from the direction in which the microphone device of this example is to be created, that is, from the direction in which the desired sound is to be obtained in order to obtain the highest sensitivity. It is assumed that no other signal is incident or can be ignored.
In this state, the configuration of the preceding stage of the adaptive noise canceller 10 becomes the adaptive noise canceller 20 that acts to cancel the identification signal as noise as shown in FIG.

That is, the adaptive noise canceller 20
Is composed of a subtraction circuit 15, an FIR filter circuit 21, and a filter coefficient operation supply circuit 22.
The output of the / D converter 14 and the reference input are the output of the A / D converter 13, and the residual output is the output of the subtraction circuit 15. When an adaptive operation is performed in the adaptive noise canceller 20 in this state, the filter coefficient calculation supply circuit 22 updates the filter coefficient of the FIR filter circuit 21 so that the output power of the subtraction circuit 15 becomes zero.

The adaptive noise canceller 20 continues the adaptive processing operation for a predetermined time such that the adaptive processing converges, and after the predetermined time has elapsed, switches SW1 and SW
Turn 2 off. This switch-off can be performed automatically using, for example, a timer. The filter coefficient operation supply circuit 22 stores the filter coefficient at the time of convergence in its memory. As this memory, a nonvolatile memory including a battery backup RAM can be used.

Therefore, thereafter, the filter coefficient of the FIR filter circuit 21 is fixed to a filter coefficient such that the output of the subtraction circuit 15 becomes zero with respect to the incidence of the identification signal.

[0046] This means that the directional characteristics of the output of the subtraction circuit 15, the direction of arrival of the identification signal, i.e., indicates that the sensitivity of the desired sound arrival direction is turned oriented characteristic becomes zero . In other words, equal to the sensitivity of the desired sound arrival direction is picked up by the smallest oriented characteristics of the microphone.

In the configuration of FIG. 1, the output of the subtraction circuit 15 is supplied to the reference input terminal 2 of the adaptive noise canceller 10 as a reference input, and the digital signal of the output of the first microphone 11 from the A / D converter 13 is output. Since the main input is supplied to the main input terminal 1 of the adaptive noise canceller 10, the amount of the desired sound leaking to the reference input is reduced, and the desired sound is prevented from deteriorating. The signal can be adaptively canceled.

Further, in the example of FIG. 1, the identification signal is input from the desired voice direction, and the switches SW1, SW2
Only by performing an operation of turning on and off the adaptive noise canceling with an arbitrary direction as a desired sound direction, it is possible to realize a super-directional microphone device in which the desired sound direction is a directional direction.

In the example shown in FIG. 1, two omnidirectional microphones are used to
Since unidirectionality is created as a reference input, variations in characteristics and sensitivity differences are small and the cost is low as compared with the case where an omnidirectional microphone and a unidirectional microphone are used.

Although the same non-directional microphone is used, there is always a certain degree of variation in characteristics or sensitivity due to the influence of the housing and the like.
However, in the above example, since the characteristic difference and the sensitivity difference between the two microphones 11 and 12 are also corrected together, it is not necessary to consider the variation in the characteristics of the microphone units.

Incidentally, in the example of FIG.
The output of No. 5 has a characteristic difference from the output of the single microphone 11 in terms of frequency, phase, and the like. The adaptive noise canceller 10 compensates for this characteristic difference of the reference input, forms a signal that approximates an unnecessary signal (noise signal) in the main input, and works to remove the unnecessary signal. As compared with the case where is a signal from a microphone having a single directivity, the burden on the adaptive filter circuit 5 is increased by the compensation operation for the characteristic difference.

The example of FIG. 3 takes this point into consideration, and the output signal of the subtraction circuit 15 is supplied to the reference input terminal 2 of the adaptive noise canceller 10 via the equalizer circuit 23 for compensating for the characteristic difference. You. Other configurations are exactly the same as those in FIG. Note that the equalizer circuit 23 can be configured by a digital filter.

According to this example, the characteristic differences such as frequency, phase, and amplitude are compensated by the equalizer circuit 23.
Accordingly, the load on the adaptive filter circuit 5 of the adaptive noise canceller 10 is reduced, the adaptive noise reduction operation is performed favorably, and a highly accurate super-directional microphone device can be realized.

When the desired sound direction is determined, for example, as in a camera-integrated VTR, the FIR filter circuit 21 is configured to supply a filter coefficient from, for example, a non-volatile memory. In the manufacturing process, as shown in FIG. 2, the identification signal is applied from the desired voice direction, and the adaptive operation is performed to obtain the optimum filter coefficient, and the obtained optimum filter coefficient is stored in the nonvolatile memory. Thus, a desired microphone device can be realized.

When the filter coefficient calculation supply circuit 22 is separated into a filter coefficient register and a filter coefficient calculation circuit, the filter coefficient calculation circuit of the filter coefficient calculation circuit 22 can be constituted by a microcomputer as described above. As described above, the microcomputer constituting the adaptive noise canceller 10 can have the role of the filter coefficient calculation circuit. As described above, the filter coefficient calculation circuit of the filter coefficient calculation supply circuit 22 operates before the use of the microphone device or operates only in the manufacturing process. No burden.

The algorithm for updating the filter coefficient is not limited to the above-mentioned LMS method, and it goes without saying that, for example, a learning identification method or another algorithm can be used.

[0057]

As described above, according to the present invention, the outputs of the two microphones are combined via the filter circuit, and the filter coefficients of the filter circuit are set so that the desired sound direction has the minimum sensitivity. By deciding,
An optimum unidirectional output signal can be obtained as a reference input. Therefore, it is possible to suppress the leakage of the desired sound to the reference input, and realize a highly accurate super-directional microphone device with less deterioration of the desired sound signal.

[Brief description of the drawings]

FIG. 1 is a block diagram of an embodiment of a microphone device according to the present invention.

FIG. 2 is a diagram for explaining an operation of a main part of the example of FIG. 1;

FIG. 3 is a block diagram of another embodiment of the microphone device according to the present invention.

FIG. 4 is a diagram illustrating an overview of an adaptive noise canceller.

FIG. 5 FIG. 4 is a diagram illustrating an example of a directional characteristic of a microphone.
You.

FIG. 6 is a block diagram illustrating an example of an adaptive filter circuit.

[Explanation of symbols]

 Reference Signs List 11 first microphone 12 second microphone 15 subtraction circuit 21 FIR filter circuit 22 filter coefficient calculation supply circuit 23 equalizer circuit 100 adaptive linear combiner 110 filter coefficient calculation means

────────────────────────────────────────────────── ─── Continued on the front page (72) Inventor Kaoru Gyokutoku Sony Corporation 7-35 Kita-Shinagawa, Shinagawa-ku, Tokyo Inside Sony Corporation (56) References JP-A-4-184400 (JP, A) JP-A Sho 61-150497 (JP, A) (58) Fields investigated (Int. Cl. ⁷ , DB name) H04R 3/00 320 H03H 21/00 H04R 1/40 320

Claims (2)

(57) [Claims] 1. A first and second microphone, filter means to which an output signal of the first microphone is supplied, and subtraction means for subtracting an output signal of the filter means from an output signal of the second microphone. And an adaptive noise canceller that adaptively reduces a noise component included in the main input signal based on the reference input signal. The characteristic of the filter means is the maximum directivity to be created .
When the identification signal is input from the degree direction and the sound is picked up by the first and second microphones, the characteristic is a result of identification so that the output of the subtraction means becomes zero. An output signal is supplied to the adaptive noise canceller as the main input signal,
Microphone and wherein the output signal of the subtraction means Ru is supplied to the adaptive noise canceler as reference input. 2. The microphone device according to claim 1, wherein the first direction in the direction of maximum sensitivity of the directivity to be created is set.
Characteristic difference between the output of the microphone and the output of the subtraction means
Comprising an equalizer for compensating the microphone device output of the subtraction means, characterized in that it is supplied to the adaptive noise canceler as reference input through the equalizer.