JP2973624B2 - Noise reduction headphone 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 noise reduction headphone device which is preferably used in a place where external noise is severe.

[0002]

2. Description of the Related Art Generally, in a headphone device for listening to audio signals such as voice and music, when used in a place where external noise is severe, hearing (hearing) of voice and music is hindered due to mixed noise. Problem. The following technologies are known to solve such a problem.

The first technique is to change the frequency characteristics of a voice or music signal to make noise less noticeable.
This technique not only has a risk of damaging the original voice and music, but also has the disadvantage of being effective only for specific noise.

In the second technique, as shown in FIGS. 4 and 5, external noise is picked up by a microphone 2 attached to the outside of a headphone housing 1, and the picked-up signal is sent to an equalizer 3. After the equalizer 3 adjusts the signal into a signal having the same amplitude and opposite phase with respect to the noise mixed in the headphone 1, the signal is sent to the headphone (speaker) 5 via the adder 4 and reproduced. The noise arriving at the human ear (ear canal) 6 is canceled. The adder 4 is supplied with a sound signal, a music signal, or the like to be heard from the signal input terminal 7.

FIG. 5 is a block diagram showing the configuration of FIG. 4 using a transfer function. External noise (noise) N from an input terminal 11 is converted by a microphone 2 having a transfer function of M into an acoustic-electric conversion. Then, the transfer function is subjected to electro-acoustic conversion by a headphone (speaker) 5 having a transfer function of −γ via an equalizer 3 having a transfer function of −γ and sent to an adder 12. This adder 1
2, the external noise (noise) N is supplied through the headphone housing 1 having the sound insulation characteristic F, is added and mixed in the headphone internal space, and reaches the human ear (ear canal 6) via the output terminal 13. I do.

In such a configuration, the sum output from the output terminal 13 is given by N ・ F + N ・ M ・ (−γ) ・ H = N (FＭM ・ γ ・ H). If −γ is adjusted so that γ = F / (H · M), FM−γ · H = 0, and
Noise can be canceled. Although not shown in FIG. 5, the electrical signal of the voice or music to be heard is supplied between the equalizer 3 and the headphone (speaker) 5 and is superimposed. According to this, there is an advantage that the original voice and music are not affected, and in principle it does not depend on the nature of noise.

However, in practice, if the adjustment by the equalizer 3 is not performed accurately, the noise canceling effect is weak, and even after the adjustment, the shape of the ear canal and the manner of wearing the headphones may cause the headphone (speaker) 5 to lose its noise. The characteristics (mainly, electro-acoustic conversion characteristics) H may change and the effect may be reduced.

Next, as a third technique, as shown in FIGS. 6 and 7, the equalizer 3 having the configuration shown in FIGS.
Has been replaced with an adaptive filter 16.
That is, in FIG. 6, a first microphone 14 for noise collection provided outside the headphone housing 1 and near the ear when the headphones are worn, and when the headphones are worn inside the headphone housing 1. A second provided at a position between the headphone (speaker) 5 and the ear canal;
The adaptive filter 16 filters the input from the first microphone 14 and sends it to the adder 4, and also transmits the filter characteristics (transfer function −γ) at this time to the second microphone 15. Is adaptively controlled so as to minimize the input from. FIG. 7 is a block diagram showing the transfer function of each unit using the transfer function of the first microphone 14 as M ₁ and the transfer function of the second microphone 15 as M ₂ . The other configuration is the same as in FIG.
Since the configuration is the same as that of FIG. 5, the corresponding portions are denoted by the same reference symbols and description thereof is omitted. In addition, a signal such as voice or music to be listened to has no correlation with noise to be reduced or canceled and does not affect the noise reduction operation, so that the description is omitted.

According to this configuration, the signal from the adder 4 is sent to the headphone (speaker) 5 to reduce the noise component, and the sound inside the headphone with the reduced noise is collected by the second microphone 15. The residual signal is sent to the adaptive filter 16 as a so-called residual, and the adaptive filter 16 learns to minimize the power of the residual signal and adjusts its own filter characteristic.

The internal configuration of the adaptive filter 16 is shown in FIG.
As shown in the figure, the filter unit 21 and the adaptive algorithm unit 2
It consists of two. An input from the first microphone 14 is supplied to a terminal 17, and is referred to as an input terminal 16 a of the adaptive filter 16 as a so-called reference input x.
It is sent to the filter unit 21 and the adaptive algorithm unit 22. The output y from the filter section 21 is taken out as the output of the adaptive filter 16 via the terminal 16 b and sent to the adder 18. The adder 18 represents the acoustic mixing inside the headphones, and the external noise (noise)
Is transmitted to an adder 18 via a terminal 19 via a terminal 19 via a sound insulation characteristic function block such as a headphone housing, and the adaptive filter output y is subtracted from the noise component d. A so-called noise residual e (= d−y) is obtained. The residual e from the adder 18 is collected by the second microphone 15 and supplied to the adaptive algorithm unit 22 via the terminal 16c. The adaptive algorithm unit 22 includes a filter unit 21
The output y obtained by filtering the input x is obtained by changing the filter coefficient of
, Adaptive control is performed to minimize the power of the residual e.

According to the third technique, since the adjustment is performed by learning, the adjustment is accurate, the noise reduction effect can be enhanced, and the readjustment is performed even if the characteristics of the headphones change later. There is an advantage that it is easy. Therefore, of the three techniques described above, the third technique is considered to be the most effective.

[0012]

However, in the noise reduction using the adaptive filter described as the third technique, there is a problem in terms of a timing shift when the successive adaptive algorithm is used. This problem will be described below.

FIG. 9 shows the internal configuration of the adaptive filter 16, and shows a specific example in which a so-called FIR (finite impulse response) filter is used as the filter unit 21. In FIG. 9, the reference input x from the input terminal 16a is composed of delay elements 23 ₁ , 23 ₂ ,.
, Sent to a 23 _L series circuit. Input terminal 16a
Input x ₀ and the delay element 23 from _1, 23 _2, ...
-, each output x ₁ from the _{23 L, x 2, ···,} x L is,
The coefficient multipliers 24 ₀ , 24 ₁ , 24 ₂ ,.
24 _L , respectively, and filter coefficients w ₀ , w ₁ , w
_2, ..., is multiplied by w _L, it is sent to the adder 25. Each filter coefficient w ₀ , w ₁ , w ₂ ,..., W _L
Is corrected by the coefficient correction signal from the adaptive algorithm unit 22, and the output y from the adder 25 is extracted from the output terminal 16b.

As the adaptive algorithm used in the adaptive algorithm unit 22, many methods have been proposed. One specific example is an LMS (Least Mean Square, Least Mean
The square algorithm will be described.

The input data and the delay elements 23 ₁ in the k-th sample period time of the data sequence of the input x (time k), 23 _2, ···, each delay output data from the 23 _L, respectively x _k0 , X _k1 , x _k2 ,..., X _kL , the input vector X to be FIR filtered
a _k, put the X _k = _{[x k0 x k1 x k2 ··· x} kL ] ^T ··· (1). T in the equation (1) indicates a transposed symbol. With respect to this input vector X _k , if the above filter coefficients (weighting coefficients) are w _k0 , w _k1 , w _k2 ,..., W _kL and the FIR filter output is y _k , (2)
It looks like an expression.

Y _k = w _k0 x _k0 + w _k1 x _k1 +... + W _kL x _kL (2) Further, a filter coefficient vector (weight vector) W
_{If k} is _defined as W _k = [w _k0 w _k1 w _k2 ... w _kL ] ^T (3), the input / output relationship is y _k = X _k ^T W _k (4) It is described as follows. If the desired response and d _k, the error epsilon _k and the output is expressed as _{ε k = d k -y k =} d k -X k T W k ··· (5). Using these, the LMS algorithm is expressed as follows: W _{k + 1} = W _k +2 με _k X _k (6) Μ in the equation (6) is a gain factor that determines the speed and stability of adaptation.

When the adaptive filter 16 in which the LMS algorithm is an adaptive algorithm is applied to the configurations shown in FIGS. 6 and 7, the result is shown in FIG. In FIG. 10, an adaptive algorithm is used because of a time delay (delay) that occurs between reproduction (electric-acoustic conversion) by the headphones (speaker) 5 and collection (acoustic-electric conversion) by the microphone 15. The residual input to the unit 22 has no correlation with the reference input supplied to the filter unit 21 at the same time, and the condition of the above equation (6) is not satisfied. That is, when the delay time of the headphones 5 is d ₁ and the delay time of the microphone 15 is d ₂ , the reference input x _k supplied to the filter unit 21 is input to the adaptive algorithm unit 22. The residuals to be processed are the delay times d ₁ and d ₁ , for example, e _{k-d1-d2.
It is two} minutes late.

As described above, if the timing of each input supplied to the adaptive filter is shifted, the condition of the above equation (6) is not satisfied, so that effective noise reduction cannot be performed. There is a disadvantage that the update type adaptive algorithm cannot be used. That is, since the amount of calculation is small, it is not possible to use a sequential adaptive algorithm suitable for practical use by an LSI or the like.

The present invention has been made in view of such circumstances, and in a noise reduction method using an adaptive filter, a so-called LMS algorithm, a sequential adaptive algorithm such as a learning identification method or the like can be used. An object is to provide a noise reduction headphone device.

[0020]

A noise reduction headphone device according to the present invention comprises: first sound-electricity conversion means provided near the ear when headphones are worn to pick up external noise;
Electro-acoustic conversion means for outputting sound to the ear canal when the headphones are mounted, second sound-electric conversion means located between the electro-acoustic conversion means and the ear canal when the headphones are mounted, and the second sound First adaptive filter means for adaptively filtering and outputting the input from the first acoustic-electrical conversion means so as to minimize the power of the input from the electric conversion means, and the first adaptive filter means; A second adaptively filtering and outputting such that the residual power between the output from the filter unit to which the output of the filter means is supplied and the output from the second acoustic-electrical conversion means is minimized. An adaptive filter means, and a filter means to which a filter coefficient of the second adaptive filter means is provided and which filters an input from the first acoustic-electrical conversion means and sends the filtered input to the first adaptive filter means. By Rukoto solves the problems described above.

Here, a microphone can be used for the first and second sound-electric conversion means, and a speaker can be used for the electric-acoustic conversion means. The adaptive filter unit includes a filter unit and an adaptive algorithm unit, and an output from the filter unit is sent to an adaptive algorithm unit of the first adaptive filter unit,
The output from the filter unit of the adaptive filter means is the second
And the output from the second sound-to-electric conversion means is sent to the adaptive algorithm section of the first adaptive filter means.

[0022]

The characteristics of the filter section of the second adaptive filter means are similar to the characteristics through the electro-acoustic conversion means and the second sound-electric conversion means, particularly the delay, and the coefficient of this filter section is By being set as the coefficient of the filter means, the characteristic of the filter means also approximates the delay. Therefore, the first
As for the reference input and the residual input to the adaptive algorithm unit of the adaptive filter means, the output (reference input) from the filter means and the output (residual) from the second acoustic-electric conversion means have the same delay. Amount, and effective adaptive processing can be realized.

[0023]

FIG. 1 is a block circuit diagram showing a configuration example of an adaptive filter used in a noise reduction headphone device according to an embodiment of the present invention. This FIG. 1 corresponds to FIG.
8 shows an example of a configuration that is used in place of the adaptive filter 16 of the noise reduction headphone device as shown in FIG. 7, and each terminal 16a, 16b and 16c is the terminal 16a of the adaptive filter 16 of FIG. 16b and 16
c respectively.

In FIG. 1, the reference input x from the terminal 16a is supplied to the filter unit 3 of the first adaptive filter 30.
1 and a delay compensation filter 51 described later, and the output from the filter 51 is sent to the adaptive algorithm unit 32 of the first adaptive filter 30. The adaptive algorithm unit 32 is supplied with the residual e from the terminal 16c, and modifies the filter coefficients of the filter unit 31 so as to minimize the power of the residual e. The output y from the filter unit 31 is
The signal is extracted through the terminal 16b, and the filter unit 41 and the adaptive algorithm unit 4 of the second adaptive filter 40 are extracted.
2 respectively. The output from the filter unit 41 is sent to the adder 52 as a subtraction signal,
The output from the adder 52 is subtracted from the residual e from c and sent to the adaptive algorithm unit 42. The adaptive algorithm unit 42 adjusts the filter coefficient of the filter unit 41 so as to minimize the power of the output from the adder 52. The filter coefficient of the filter section 41 is sent to the delay compensation filter 51 and copied, so that the same characteristic (particularly, delay characteristic) is realized.

Since other configurations are the same as those of the noise reduction headphone device shown in FIGS. 6 and 7 and the like, they are not shown, but the reference input terminal 16a is provided near the ear when the headphones are mounted. An input from a microphone, which is a first acoustic-electric conversion means for collecting external noise (noise), is supplied, and an output from an output terminal 16b is provided near the ear when headphones are worn. To the headphones (speakers), which are electro-acoustic conversion means for outputting sound to the ear canal.
6c is supplied with an input from a microphone, which is a second sound-to-electricity conversion means, located between the headphone (speaker) and the ear canal when the headphone is worn. Further, the specific internal configuration of each of the first and second adaptive filters 30 and 40 may be the configuration shown in FIGS. 8 and 9 described above, and the filter 51 is the same as the filter 21 of FIG. An FIR filter configuration may be used. Here, the filter unit 41 and the filter 51 of the adaptive filter 40 use the same filter structure, and the same characteristic (particularly, the same delay characteristic) can be realized by copying the filter coefficient.

Next, the operation will be described with reference to FIGS. The operation of this embodiment is similar to the operation of the adaptive filter 30 described above.
The operation centered on the adaptive algorithm unit 31 of the adaptive filter 40 and the operation centered on the adaptive algorithm unit 41 of the adaptive filter 40 can be roughly classified. In the following description, it is assumed that the adaptation process is first performed by the adaptation algorithm unit 41 (FIG. 2), and the operation of the adaptation algorithm unit 31 is performed after the completion of the process (FIG. 3). However, it is also possible to make these operations proceed simultaneously by appropriately setting the conditions.

In FIGS. 2 and 3, similarly to the example of FIG. 7 described above, the sound insulation characteristic of the headphone housing 1 is represented by a transfer function F. first acoustic provided headphones outside to pick up noise) - the transfer function of the microphone 14 is an electrical conversion section and M _1, is provided in the vicinity of the ear when wearing headphones, outputs sound to the ear canal Electricity to do
The transfer function of the headphone (speaker) 5 as the sound conversion means is set to H, and the second sound-electric conversion means provided inside the headphone and located between the headphone (speaker) 5 and the ear canal when the headphone is attached. The transfer function of the microphone 15 is represented by M _2, and the acoustic mixing inside the headphones is represented as addition by the adder 12.

That is, in FIGS. 2 and 3, the external noise (noise) N from the input terminal 11 is subjected to acoustic-electric conversion by the microphone 14 having the transfer function M ₁ and supplied to the filter unit 31 and the filter 51. At the same time, the output from the filter unit 31 is subjected to electro-acoustic conversion by the headphones (speaker) 5 having the transfer function H and sent to the adder 12. The adder 12, the external noise N is supplied through the headphone case 1 sound insulating properties F are added mixed with headphones interior space, as well as reach the human ear, the microphone 15 of the transfer function M ₂ The signal is subjected to acoustic-electric conversion and sent to the adaptive algorithm unit 32 and the adder 52. It should be noted that voices, music signals, and the like that are originally intended to be reproduced by the headphones (the user intends to listen to them) have no correlation with the noise to be reduced and do not affect the noise reduction operation. doing.

In FIG. 2, a certain kind of noise is reproduced by the headphone 5 and collected by the microphone 15, and the same kind of noise is supplied to the filter unit 41 and the adaptive algorithm unit 42. As this kind of noise, noise collected by the microphone 14 may be used,
It is preferable to use noise generated from a white noise generator or the like. The residual obtained by subtracting the output of the filter unit 41 from the signal collected by the microphone 15 is input to the adaptive algorithm unit 42. The adaptive algorithm unit 42
The coefficient of the filter unit 41 is modified by learning so as to minimize the power of the residual. As a result, the filter unit 41
Is an approximation of the characteristic through the headphone 5 and the microphone 15, particularly, the above-described time delay. Each filter coefficient of the filter unit 41 is copied as each filter coefficient of the filter 51. That is,
The filter unit 41 and the filter 51 have, for example, an FIR filter configuration with the same number of taps, and the same characteristic (delay characteristic) can be realized by copying each coefficient of the filter unit 41 to the filter 51.

Next, in FIG. 3, the noise signal picked up by the microphone 14 is filtered by the filter unit 2 into pseudo noise, and thereafter, the headphone (speaker) thereof.
5 and reproduced. The reproduced pseudo noise acoustically cancels out the noise mixed in the headphones, and the residual is collected by the microphone 15 and input to the adaptive algorithm unit 32. This residual signal is
Although the delay is delayed by the delay in the headphone 5 and the microphone 15, this delay is realized by the filter 51, so that the noise signal supplied as a reference input to the adaptive algorithm unit 32 has the same delay. .

Therefore, when the LMS algorithm as described above is used for the adaptive algorithm unit 32, the requirement of the above-mentioned expression (6) is satisfied, and effective adaptive processing is performed. Therefore, by using a sequential adaptive algorithm that requires a small amount of calculation such as an LMS algorithm or a learning identification method, it can be put to practical use by an LSI or the like, and can be further put to practical use.

The present invention is not limited to the above embodiment. For example, the successive adaptive algorithm is not limited to the above-mentioned LMS method, and various other successive adaptive algorithms can be used.

[0033]

As is clear from the above description, according to the noise reduction headphone device of the present invention, the input from the first acoustic-electric conversion means for picking up external noise is adaptively filtered. A first adaptive filter means for outputting a sound, an output from the first adaptive filter means, an electro-acoustic conversion means for supplying a sound to an ear canal when headphones are mounted, and the electro-sound when the headphones are mounted. A second sound-to-electricity conversion means located between the conversion means and the ear canal, an output from a filter section to which the output of the first adaptive filter means is supplied, and a signal from the second sound-to-electricity conversion means. Second adaptive filter means for adaptively performing filter processing so as to minimize the power of the residual with respect to the output of the second adaptive filter means; Filter means for filtering an input from the electric conversion means and sending the processed signal to the first adaptive filter means (algorithm part thereof), wherein the first adaptive filter means is provided with the second sound-electric conversion means. The filter coefficients are modified so as to minimize the power of the input from. For this reason, the characteristics of the filter section of the second adaptive filter means are similar to the characteristics through the electro-acoustic conversion means and the second sound-electric conversion means, particularly the delay, and the coefficient of this filter section is By being set as a coefficient of the filter means,
The characteristic of the filter means is also an approximation of the delay, and the reference input and the residual input to the adaptive algorithm unit of the first adaptive filter means are the output (reference input) from the filter means and the second input. And the output (residual) from the sound-electricity conversion means has the same delay amount, and effective adaptive processing can be realized by the successive adaptive algorithm.

Therefore, the use of a sequential adaptive algorithm requiring a small amount of calculation, such as the LMS algorithm or the learning identification method, makes it possible to use the LSI or the like, and further facilitates the practical use.

[Brief description of the drawings]

FIG. 1 is a block circuit diagram showing a main configuration of a noise reduction headphone device according to an embodiment of the present invention.

FIG. 2 is a block diagram for explaining the operation of the embodiment.

FIG. 3 is a block diagram for explaining the operation of the embodiment.

FIG. 4 is a block diagram showing a conventional example of a noise reduction headphone device.

FIG. 5 is a block diagram for explaining the operation of the conventional example.

FIG. 6 is a block diagram showing another conventional example of a noise reduction headphone device.

FIG. 7 is a block diagram for explaining the operation of the other conventional example.

FIG. 8 is a block circuit diagram showing an example of an internal configuration of an adaptive filter.

FIG. 9 is a block circuit diagram showing a specific example of the internal configuration of the adaptive filter.

FIG. 10 is a block diagram for explaining the operation of the other conventional example.

[Explanation of symbols]

 1 ····· Headphone housing (sound insulation characteristic part) 5 ··········· Headphone (speaker) 11 ····· Noise input terminal 12 ··· (acoustic) adder 14 ··· ··· (first) microphone 15 ··· (second) microphone 16 ··· Adaptive filters 21, 31, 41 ··· Filter units 22, 32, 42 ··· Adaptive algorithm unit 30 (first) adaptive filter 40 (second) adaptive filter

────────────────────────────────────────────────── ─── Continued on the front page (72) Inventor Takeshi Hara 6-7-35 Kita-Shinagawa, Shinagawa-ku, Tokyo Inside Sony Corporation (58) Field surveyed (Int.Cl. ⁶ , DB name) H04R 1 / 10 101 H04R 3/04 101 G10K 11/16

Claims (1)

(57) [Claims] A first sound-electricity conversion means provided near the ear when the headphones are worn to pick up external noise; and an electric sound-electricity means provided near the ears when the headphones are worn and outputting sound to the ear canal. Sound conversion means, second sound-to-electricity conversion means located between the electric-to-sound conversion means and the ear canal when headphones are worn, and minimizing the power of the input from the second sound-to-electricity conversion means. A first adaptive filter means for adaptively filtering and outputting an input from the first acoustic-electrical conversion means, and a filter unit to which an output from the first adaptive filter means is supplied. A second adaptive filter means for adaptively filtering and outputting the residual power between the output of the filter unit and the output from the second sound-electricity conversion means so as to minimize the residual power; This second adaptive filter A noise reduction headphone device, comprising: a filter unit to which a filter coefficient of the filter unit is given and an input from the first acoustic-electric conversion unit to be filtered and sent to the first adaptive filter unit. .