JP5811993B2 - Headphones, headphone noise reduction method, noise reduction processing program - Google Patents

Headphones, headphone noise reduction method, noise reduction processing program Download PDF

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JP5811993B2
JP5811993B2 JP2012267956A JP2012267956A JP5811993B2 JP 5811993 B2 JP5811993 B2 JP 5811993B2 JP 2012267956 A JP2012267956 A JP 2012267956A JP 2012267956 A JP2012267956 A JP 2012267956A JP 5811993 B2 JP5811993 B2 JP 5811993B2
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noise reduction
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宏平 浅田
宏平 浅田
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ソニー株式会社
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The present invention relates to a headphone, a headphone noise reduction method, and a noise reduction processing program .

Japanese Patent No. 2778173 Japanese Patent No. 2867461

  With the widespread use of portable audio players, good reproduction with reduced external noise to listeners by reducing external environmental noise for headphones and earphones for portable audio players Noise reduction systems that provide sound field space are beginning to become popular.

  An example of this type of noise reduction system is an active noise reduction system that performs active noise reduction, and basically includes the following configuration. That is, external noise (noise) is picked up by a microphone as an acoustic-electric conversion means, and a noise-reduced voice signal that is acoustically opposite in phase to the noise is generated from the picked-up noise signal. The generated noise-reduced audio signal is acoustically reproduced by a speaker as an electro-acoustic conversion means and is acoustically synthesized with the noise to reduce the noise (Patent Document 1 (Japanese Patent No. 2778173). Publication))).

  In this active type noise reduction system, conventionally, the part that generates the noise-reduced audio signal is composed of an analog circuit (analog filter), so that it can reduce noise appropriately in any noise environment. It is fixed to a simple filter circuit.

  In addition, a headphone device equipped with a noise reduction system that uses an adaptive filter that uses adaptive processing can be used to play music while reducing the noise even in environments where there is a large amount of external noise. It has been proposed (see, for example, Patent Document 2 (Japanese Patent No. 2867461)).

  The noise reduction system for noise reduction headphones described in Patent Document 2 automatically sets an adaptive filter to an optimum one using adaptive signal processing, and collects external noise outside the headphone housing. And a microphone for collecting a residual (error) component as a result of acoustic synthesis by adaptive signal processing is provided inside the headphone housing.

  In this adaptive processing noise reduction system, the residual signal from the microphone provided inside the headphone housing is analyzed, and the adaptive filter is updated to adaptively reduce noise with respect to external noise. It is configured as follows.

  By the way, in general, the noise environment characteristic varies greatly depending on the environment of the place such as an airfield, a station platform, a factory, etc., even if observed with a frequency characteristic. Therefore, it is originally desirable to use an optimum filter characteristic for noise reduction according to each noise environment characteristic.

  However, as described above, the conventional active noise reduction system is fixed to a filter circuit having a single filter characteristic that can reduce noise appropriately in any noise environment. For this reason, the conventional active noise reduction system has a problem that noise reduction suitable for the noise environment characteristics of the place where noise reduction is to be performed cannot be performed.

  Therefore, it is conceivable that a plurality of filter circuits having various filter characteristics are provided instead of a filter circuit having a single filter characteristic, and a filter circuit suitable for a noise environment characteristic of a place is switched and selected. In this case, the hardware circuit itself is switched because of the conventional analog circuit configuration.

  However, when a plurality of filter circuits are provided in this manner and one of them is switched and selected, there is a problem that the hardware configuration becomes large and the cost increases, and the portable device It is not practical as a noise reduction system used for the above.

  On the other hand, if a noise reduction system using adaptive processing is used, the adaptive filter is updated adaptively to the noise at the place where the noise reduction system is used, so there is no need to provide a plurality of filter circuits.

  For this reason, a number of techniques for reducing (cancelling) noise using adaptive signal processing have been proposed in patent literature and conference presentations, but the stability problem as a system, the processing scale increases, and the target is periodic. The current situation is that it is not suitable for noise waveforms, and problems such as cost effectiveness (cost performance) have not been cleared and are not actually commercialized.

  In view of the above points, an object of the present invention is to provide a noise reduction apparatus that can employ an active noise reduction system that does not use adaptive processing, but can perform noise reduction appropriately corresponding to a noise environment. And

In order to solve the above problem, the headphone according to claim 1 is a digital microphone that is installed toward the outside of the headphone housing and picks up external noise of the headphone, and an analog audio signal picked up by the microphone. A / D conversion means for converting to an audio signal, muting means for muting a predetermined input audio signal, and when the predetermined input audio signal is muted by the muting means, the A / D Noise characteristics analyzing means for analyzing noise characteristics after performing high-frequency component removal and downsampling on the digital audio signal obtained by the converting means, and a plurality of sets prepared in advance according to a plurality of types of noise characteristics holding means for holding the parameters of, among the multiple sets of parameters stored in said holding means, said Neu A selection setting means for selecting a set of parameters suitable for noise reduction of the noise characteristics of the analysis result by the characteristics analysis means, by using a set of the digital audio signal with the selected parameters, generating a noise reducing audio signal comprising a digital processing means for, with the noise reducing audio signal, and a speaker for acoustically reproducing said predetermined input sound signal.

The headphones configured as described above perform active noise reduction, but the noise-reduced audio signal is generated by digital processing means. The holding means holds a plurality of parameters corresponding to the characteristics of the noise corresponding to various noise environments, and the digital processing means sets an appropriate noise characteristic parameter among the plurality of parameters. It can be used to generate a noise reduced audio signal. Therefore, it is possible to reduce noise appropriately corresponding to various noise environments.

In this case, the hardware configuration includes selection setting means for holding a plurality of parameters corresponding to a plurality of types of noise characteristics in the holding means, and selecting parameters suitable for noise reduction among them. Therefore, it does not become large-scale compared with the case where an analog filter circuit is used. That is, even when dealing with a wide variety of noise characteristics, it is only necessary to hold a plurality of parameters corresponding to the various noise characteristics, so a large number of analog filter circuits are provided and switched. Compared to the case, the configuration is simple and the cost is advantageous.

According to the present invention, even if an active noise reduction method is used, it is possible to perform noise reduction appropriately corresponding to the noise environment, and the circuit scale is not increased, and the cost is practical. A headphone equipped with a noise reduction device can be realized.

1 is a block diagram of an example of a headphone device to which a first embodiment of a noise reduction device according to the present invention is applied; It is the figure which showed the structure of 1st Embodiment of the noise reduction apparatus by this invention using the transfer function. It is a figure used in order to demonstrate embodiment of the noise reduction apparatus by this invention. It is a figure used in order to explain a 1st embodiment of a noise reduction device by this invention. It is a figure which shows the flowchart for demonstrating operation | movement of the principal part in embodiment of the noise reduction apparatus by this invention. It is a figure used in order to demonstrate embodiment of the noise reduction apparatus by this invention. It is a block diagram of the example of the headphone apparatus with which 2nd Embodiment of the noise reduction apparatus by this invention was applied. It is the figure which showed the structure of 2nd Embodiment of the noise reduction apparatus by this invention using the transfer function. It is a figure used in order to explain the attenuation characteristic of a noise reduction system of a feedback system and a noise reduction system of a feedforward system. It is a figure used in order to demonstrate 3rd and 4th embodiment. It is a figure used in order to demonstrate 3rd and 4th embodiment. It is a figure used in order to demonstrate 3rd and 4th embodiment. It is a figure used in order to demonstrate 3rd and 4th embodiment. It is a block diagram of the example of the headphone apparatus with which 3rd Embodiment of the noise reduction apparatus by this invention was applied. It is a figure used in order to demonstrate the characteristic of 3rd Embodiment of the noise reduction apparatus by this invention. It is a block diagram of the example of the headphone apparatus with which 4th Embodiment of the noise reduction apparatus by this invention was applied. It is a block diagram of the example of the headphone apparatus with which 5th Embodiment of the noise reduction apparatus by this invention was applied. It is a block diagram of the other example of the headphone apparatus with which 5th Embodiment of the noise reduction apparatus by this invention was applied. It is a figure which shows the detailed structural example of the one part block of FIG. It is a block diagram of the example of the headphone apparatus with which 6th Embodiment of the noise reduction apparatus by this invention was applied. It is a block diagram of the example of the headphone apparatus with which 7th Embodiment of the noise reduction apparatus by this invention was applied. It is a figure which shows the flowchart for demonstrating operation | movement of the principal part of 7th Embodiment of the noise reduction apparatus by this invention. It is a figure which shows the specific structural example of the one part block of the structural example of 7th Embodiment of FIG. It is a figure which shows the specific structural example of the one part block of the structural example of 7th Embodiment of FIG. It is a figure used in order to demonstrate operation | movement of the principal part of 7th Embodiment of the noise reduction apparatus by this invention. It is a figure which shows the flowchart for demonstrating operation | movement of the principal part of 7th Embodiment of the noise reduction apparatus by this invention. It is a block diagram which shows the structural example of the headphone apparatus of 8th Embodiment. It is a figure which shows the flowchart used in order to demonstrate operation | movement of the principal part of 8th Embodiment. It is a block diagram which shows the structural example of the headphone apparatus of 9th Embodiment.

  Several embodiments of the noise reduction device according to the present invention will be described below with reference to the drawings. Each of the embodiments described below is a case where the noise reduction device according to the present invention is applied to a headphone device as an embodiment of the noise reduction sound output device according to the present invention.

  By the way, as a system for performing active noise reduction, there are a feedback system (feedback type) and a feed forward system (feed forward type). The present invention can be applied to any type of noise reduction system.

  There are two methods for changing the characteristics of the noise reduction device according to the noise environment: a manual selection method according to the user's selection instruction and an automatic change method for automatically changing the characteristics according to the noise environment. is there.

[Manual selection method]
First Embodiment (Feedback Noise Reduction Device)
First, an embodiment in which the present invention is applied to a feedback type noise reduction system will be described. FIG. 1 is a block diagram showing a configuration example of an embodiment of a headphone device to which an embodiment of a noise reduction device according to the present invention is applied.

  For the sake of simplicity, FIG. 1 shows the configuration of only the right ear side portion of a listener (listener) 1 of the headphone device. The same applies to other embodiments described later. Needless to say, the portion on the left ear side is configured in the same manner.

  FIG. 1 shows a state in which the listener 1 is mounted with the headphone device of the embodiment, so that the right ear of the listener 1 is covered with a right-ear headphone housing (housing) 2. Inside the headphone housing 2, a headphone driver unit (hereinafter simply referred to as a driver) 11 is provided as an electro-acoustic conversion means for acoustically reproducing an audio signal that is an electric signal.

  Then, for example, a music signal is supplied to the power amplifier 15 through the audio signal input terminal 12 through the equalizer circuit 13 and the adder circuit 14, and the music signal through the power amplifier 15 is supplied to the driver 11 for sound reproduction and listener. The reproduction sound of the music signal is emitted to the right ear of 1.

  The audio signal input terminal 12 is composed of a headphone plug that is plugged into the headphone jack of the portable music player. In an audio signal transmission path between the audio signal input terminal 12 and the left and right ear drivers 11, an equalizer circuit 13, an adder circuit 14, a power amplifier 15, and an acoustic-electric conversion unit, which will be described later, are provided. The noise reduction device unit 20 includes a microphone 21, a microphone amplifier (hereinafter simply referred to as a microphone amplifier) 22, a noise reduction filter circuit 23, a memory 24, a memory controller 25, an operation unit 26, and the like. .

  Although not shown, the noise reduction device 20 and the driver 11, microphone 21, and headphone plug constituting the audio signal input terminal 12 are connected by a connection cable. Reference numerals 20 a, 20 b, and 20 c are connection terminal portions to which a connection cable is connected to the noise reduction device portion 20.

  In the first embodiment of FIG. 1, in the music listening environment of the listener 1, noise that enters the music listening position of the listener 1 inside the headphone housing 2 from the noise source 3 outside the headphone housing 2 is fed back. To be able to listen to music in a good environment.

  In the feedback type noise reduction system, the noise at the acoustic listening position (noise cancellation point Pc) where the noise and the sound reproduction sound of the noise-reduced speech signal are synthesized is collected by the microphone. Sounds.

  Therefore, in the first embodiment, the noise collecting microphone 21 is provided at the noise canceling point Pc inside the headphone housing (housing) 2. Since the sound at the position of the microphone 21 serves as a control point, the noise cancellation point Pc is set at a position close to the normal ear, that is, the front surface of the diaphragm of the driver 11 in consideration of the noise attenuation effect. Provided.

  The noise-reduced phase component of the noise collected by the microphone is generated as a noise-reduced audio signal by the noise-reduced audio signal generation unit, and the generated noise-reduced audio signal is supplied to the driver 11 for sound reproduction. The noise that enters the headphone housing 2 from the outside is reduced.

  Here, the noise in the noise source 3 and the noise 3 ′ entering the headphone housing 2 are not the same characteristics. However, in the noise reduction system of the feedback system, noise 3 ′ entering the headphone housing 2, that is, noise 3 ′ to be reduced is collected by the microphone 21.

  Therefore, in the feedback method, the noise-reduced audio signal generation unit may generate a reverse phase component of the noise 3 ′ so as to cancel the noise 3 ′ picked up by the microphone 21 at the noise cancellation point Pc.

  In this embodiment, the digital filter circuit 23 is used as a noise reduction voice signal generation unit of a feedback system. In this embodiment, since the noise-reduced audio signal is generated by the feedback method, the digital filter circuit 23 is hereinafter referred to as the FB filter circuit 23.

  The FB filter circuit 23 includes a DSP (Digital Signal Processor) 232, an A / D conversion circuit 231 provided at the preceding stage, and a D / A conversion circuit 233 provided at the subsequent stage.

  The obtained analog audio signal collected by the microphone 21 is supplied to the FB filter circuit 23 through the microphone amplifier 22 and converted into a digital audio signal by the A / D conversion circuit 231. Then, the digital audio signal is supplied to the DSP 232.

  The DSP 232 is configured with a digital filter for generating a feedback-type digital noise-reduced audio signal. The digital filter generates the digital noise-reduced audio signal having a characteristic corresponding to a filter coefficient as a parameter set in the digital audio signal input thereto. In this embodiment, the filter coefficient set in the digital filter of the DSP 232 is supplied from the memory 24 through the memory controller 25.

  In this embodiment, in the memory 24, noise in a plurality of different noise environments can be reduced by a noise-reduced audio signal by a feedback method generated by a digital filter of the DSP 232, as will be described later. A plurality of (multiple sets) parameters are stored as filter coefficients.

  The memory controller 25 reads one specific (one set) filter coefficient from the memory 24 and sets it as a digital filter of the DSP 232.

  In this embodiment, the operation output signal of the operation unit 26 is supplied to the memory controller 25, and the memory controller 25 specifies from the memory 24 according to the operation output signal from the operation unit 26. One (one set) of filter coefficients is selected and read out, and set in the digital filter of the DSP 232.

  The digital filter of the DSP 232 generates a digital noise reduced audio signal corresponding to the filter coefficient that is selectively read from the memory 24 via the memory controller 25 and set as described above.

  The digital noise reduced sound signal generated by the DSP 232 is converted into an analog noise reduced sound signal by the D / A conversion circuit 233. The analog noise reduced audio signal is supplied to the adder circuit 14 as an output signal of the FB filter circuit 23.

  An input sound signal (music signal or the like) S that the listener 1 wants to listen to through headphones is supplied to the adder circuit 14 through the sound signal input terminal 12 and the equalizer circuit 13. The equalizer circuit 13 corrects the sound characteristics of the input audio signal.

  The audio signal resulting from the addition by the adder circuit 14 is supplied to the driver 11 through the power amplifier 15 and reproduced as sound. The sound reproduced and emitted by the driver 11 includes a sound reproduction component by the noise-reduced sound signal generated in the FB filter 23. The sound reproduction component and the noise 3 ′ of the noise-reduced sound signal out of the emitted sound reproduced by the driver 11 are acoustically synthesized, so that the noise 3 ′ is reduced at the noise cancellation point Pc. (Cancelled)

  The noise reduction operation of the feedback type noise reduction apparatus described above will be described with reference to FIG. 2 using a transfer function.

  That is, FIG. 2 is a block diagram showing each part using its transfer function in correspondence with the block diagram shown in FIG. In FIG. 2, A is a transfer function of the power amplifier 15, D is a transfer function of the driver 11, M is a transfer function corresponding to the parts of the microphone 21 and the microphone amplifier 22, and -β is a filter designed for feedback. It is a transfer function. H is a transfer function of the space from the driver 11 to the microphone 21, and E is a transfer function of the equalizer 13 applied to the audio signal S for listening. Each of the above transfer functions is assumed to be expressed in a complex manner.

  In FIG. 2, N is noise that has entered the vicinity of the position of the microphone 21 in the headphone housing 2 from an external noise source, and P is the sound pressure that reaches the ear of the listener 1. Note that the cause of external noise being transmitted into the headphone housing 2 is, for example, the case where the sound pressure leaks from the gap between the ear pad portions or the headphone housing 2 vibrates due to the sound pressure. 2 The case where sound is transmitted inside is considered.

  When expressed as shown in FIG. 2, the block of FIG. 2 can be expressed by (Equation 1) of FIG. When attention is paid to the noise N in (Equation 1), it can be seen that the noise N is attenuated to 1 / (1 + ADHMβ). However, in order for the system of (Equation 1) to stably operate as a noise canceling mechanism in the noise reduction target frequency band, (Equation 2) of FIG. 3 needs to be established.

  In general, the absolute value of the product of each transfer function in the noise reduction system of the feedback method is 1 or more (1 << | ADHMβ |), and the stability determination of Nyquist Nyquist in classical control theory, The stability of the system with respect to (Equation 2) in FIG. 3 can be interpreted as follows.

  In FIG. 2, an “open loop” of the transfer function (−ADHMβ) is considered by cutting one place in the loop portion (loop portion from the microphone 21 to the driver 11) related to the noise N. This has a characteristic represented by a Bode diagram as shown in FIG.

When this open loop is targeted, the condition that the above (Equation 2) is satisfied from the Nyquist stability determination is as shown in FIG.
-Phase 0 deg. The gain must be less than 0 dB when passing the point of 0. When the gain is 0 dB or more, the phase is 0 deg. This means that the two conditions must not be included.

  When the above two conditions are not satisfied, the loop is positively fed back and oscillates (howling). In FIG. 4, Pa and Pb represent phase margins, and Ga and Gb represent gain margins. If these margins are small, the risk of oscillation increases due to individual differences and variations in headphones wearing.

  Next, in addition to the noise reduction function, a case where necessary sound is reproduced from a headphone driver will be described.

  The audio signal S to be listened to in FIG. 2 is actually a music signal, a microphone sound outside the housing (used as a hearing aid function), an audio signal via communication (used as a headset), etc. It is a generic term for signals that should be reproduced by a headphone driver.

  Focusing on the signal S in (Expression 1) described above, if the equalizer E is set as shown in (Expression 3) shown in FIG. 3, the sound pressure P is expressed as (Expression 4) in FIG. Is done.

  Therefore, if the position of the microphone 21 is very close to the ear position, H is a transfer function from the driver 11 to the microphone 21 (ear), and A and D are transfer functions of the characteristics of the power amplifier 15 and the driver 11, respectively. It can be seen that the same characteristics as those of headphones having no normal noise reduction function can be obtained. At this time, the transfer characteristic E of the equalizer circuit 13 is substantially the same as the open-loop characteristic viewed from the frequency axis.

  As described above, the headphone device having the configuration shown in FIG. 1 can listen to the audio signal to be listened to without any trouble while reducing noise. However, in this case, in order to obtain a sufficient noise reduction effect, the digital filter configured by the DSP 232 has a filter coefficient corresponding to the characteristics of the noise transmitted from the external noise source 3 into the headphone housing 2. Need to be set.

  As described above, there are various noise environments in which noise is generated, and the frequency characteristics and phase characteristics of the noise correspond to the respective noise environments. For this reason, it cannot be expected that a single filter coefficient can provide a sufficient noise reduction effect in all noise environments.

  Therefore, in this embodiment, as described above, a plurality of (multiple sets) filter coefficients corresponding to various noise environments are stored and prepared in advance in the memory 24, and the plurality of filter coefficients are prepared. Then, a suitable one is selected and read out, and set to the digital filter configured in the DSP 232 of the FB filter circuit 23.

  The filter coefficient to be set in the digital filter is calculated in advance in the memory 24 so as to collect noise in each of various noise environments and reduce (cancel) the noise. It is desirable to memorize it. For example, it collects noise in various noise environments such as station platforms, airfields, trains running on the ground, subway trains, town crowds, large stores, etc., and reduces the noise (cancellation) ) Can be calculated in advance and stored in the memory 24.

  In the first embodiment, selection of an appropriate filter coefficient from a plurality (a plurality of sets) of filter coefficients stored in the memory 24 is manually performed by the user. Therefore, the operation unit 26 operated by the user is connected to the memory controller 25.

  In this embodiment, the operation unit 26 includes, for example, a non-locking type push switch as a filter coefficient changing operation unit, and the memory controller 25 reads the filter read from the memory 24 each time the listener presses the push switch. The coefficient set is changed and supplied to the FB filter circuit 23.

  FIG. 5 shows a flowchart of memory read control in the memory controller 25 in this case. That is, the memory controller 25 monitors the operation signal from the operation unit 26, and determines whether or not the push switch has been pressed and a filter coefficient changing operation instruction has been issued (step S1).

  If it is determined in step S1 that there is no filter coefficient changing operation instruction, the memory controller 25 repeats this step S1 and waits for a filter coefficient changing operation instruction. When it is determined in step S1 that there has been an instruction to change the filter coefficient, the memory controller 25 changes the set of filter coefficients read from the memory 24 to a filter coefficient in the next order different from that of the previous one, and the FB filter circuit (Step S2). Then, the process returns to step S1.

  Here, in the memory controller 25, when a reading order is determined in advance for a plurality (a plurality of sets) of filter coefficients stored in the memory 24 and it is determined that a filter coefficient changing operation instruction has been given, the reading order is determined. Accordingly, a plurality of filter coefficients are read out and changed in order and cyclically.

  For example, a set of parameters that can obtain four types of noise reduction effects represented by the “noise attenuation curve (noise attenuation characteristic)” shown in FIG. 6, that is, a set of filter coefficients, is written in the memory 24. Suppose that In the example of FIG. 6, curve characteristics that reduce noise in each case are obtained with respect to four types of noise characteristics when noise is mainly distributed in each of a low frequency range, a mid-low frequency range, a mid frequency range, and a wide frequency range. This is the case where the filter coefficients to be stored are stored in the memory 24.

  In this case, as shown in FIG. 6, when the noise is mainly distributed in the low frequency range, the filter coefficient for obtaining the noise reduction characteristic of the low frequency emphasis curve for reducing the noise is the first, and the noise is mainly distributed in the middle and low frequency range. The second filter coefficient to obtain the noise reduction characteristic of the mid-low range emphasis curve that reduces noise, the third filter coefficient to obtain the noise reduction characteristic of the mid-range emphasis curve to reduce noise when the noise is mainly distributed in the mid range, When the noise is distributed over a wide band, the filter coefficient that obtains the noise reduction characteristic of the wide-band curve that reduces the noise is assumed to be the fourth, and whenever the push switch is pressed and a filter coefficient changing operation instruction is issued, the first is performed. The filter coefficient read from the memory 24 is changed in the order of second, third, fourth, first, and so on.

  By changing the listener 1 in this way, the noise reduction effect can be confirmed with its own ears, and when the filter coefficient that is felt that the sufficient noise reduction effect has been obtained is read out, After that, stop pressing the push switch. Then, the memory controller 25 is in a state of continuously reading the filter coefficient read at that time, and is controlled to read out the filter coefficient selected by the user.

  In this case, in order for the listener to confirm the noise reduction effect more reliably, it is better to perform in an environment where the reproduced sound by the audio signal S is not emitted from the driver 11. For this purpose, in the environment where the audio signal S is not input, in addition to the method in which the listener operates the operation unit 26 to confirm the noise reduction effect, when the audio signal S is being input and being reproduced, A method of muting the audio signal to the adder circuit 14 can be adopted for a predetermined time after which the noise reduction effect can be confirmed after the push switch of the operation unit 26 is pressed.

  In the example of FIG. 6 described above, as described above, the noise in each noise environment is actually measured and the corresponding filter coefficient is not set. This is equivalent to the case where filter coefficients are set and stored in the memory 24 so as to obtain a curve characteristic that reduces noise in each case assuming a state of distribution in four types of mid-range and wide-band.

  Even with such simply set filter coefficients, according to the noise reduction apparatus of this embodiment, filter coefficients suitable for each noise environment can be selected. As compared with the case where the filter coefficient is fixedly fixed, a more effective noise reduction effect can be obtained.

  Note that the memory controller 25 in the above-described embodiment can also be configured in the DSP 232.

  Further, in the above description, the equalizer characteristic in the equalizer circuit 13 is not mentioned, but in the case of a feedback type noise reduction device, when the noise reduction curve is changed by changing the filter coefficient of the digital filter, Since the input audio signal S to be listened to is affected by the frequency curve of the noise reduction effect, it is necessary to change the equalizer characteristics in accordance with the change of the filter coefficient of the digital filter.

  Therefore, for example, the memory 24 stores parameters for changing the equalizer characteristics of the equalizer circuit 13 in correspondence with each of the plurality of filter coefficients of the digital filter, and the memory controller 25 changes the filter coefficients. A corresponding parameter is supplied to the equalizer circuit 13 to change its equalizer characteristic.

  Note that the equalizer circuit 13 may be configured in the DSP 232 as a configuration of a digital equalizer circuit. In that case, the audio signal S is converted into a digital signal and supplied to the equalizer circuit in the DSP 232. Then, the memory controller 25 may read the parameters corresponding to the change of the filter coefficient of the digital filter from the memory 24 and supply the parameters to the digital equalizer circuit to change the equalizer characteristics.

[Second Embodiment (Feedforward Noise Reduction Device)]
FIG. 7 is a configuration example of an embodiment of a headphone device to which an embodiment of a noise reduction device according to the present invention is applied, and is a block diagram showing a case where a feed forward method is used instead of the feedback method of FIG. In FIG. 7, the same parts as those in FIG.

  The noise reduction device unit 30 according to the second embodiment includes a microphone 31, a microphone amplifier 32, a noise reduction filter circuit 33, a memory 34, a memory controller 35, an operation unit 36, and the like as acoustic-electric conversion means. The reduction device unit 30 is provided.

  The noise reduction device unit 30 is connected to the driver 11, the microphone 31, and the headphone plug constituting the audio signal input terminal 12 by a connection cable, as in the above-described feedback type noise reduction device unit 20. Reference numerals 30 a, 30 b, and 30 c denote connection terminal portions to which connection cables are connected to the noise reduction device portion 30.

  In the second embodiment, in the music listening environment of the listener 1, noise that enters the music listening position of the listener 1 in the headphone housing 2 from the noise source 3 outside the headphone housing 2 is reduced by a feed forward method. So that you can listen to music in a good environment.

  As shown in FIG. 7, the feed-forward noise reduction system basically has a microphone 31 installed outside the headphone housing 2, and the microphone 31 is suitable for the noise 3 collected. The noise-reduced audio signal is generated by performing an appropriate filtering process, and the generated noise-reduced audio signal is acoustically reproduced by the driver 11 inside the headphone housing 2, and noise (noise 3) near the ear of the listener 1. ') Is canceled.

  The noise 3 picked up by the microphone 31 and the noise 3 ′ in the headphone housing 2 have different characteristics depending on the difference in spatial position between them (including the difference between the inside and outside of the headphone housing 2). Become. Therefore, in the feedforward method, a noise-reduced audio signal is generated in consideration of a difference in spatial transfer function between the noise from the noise source 3 collected by the microphone 31 and the noise 3 ′ at the noise cancellation point Pc. .

  In this embodiment, a digital filter circuit 33 is used as a noise reduction audio signal generation unit of a feedforward method. In this embodiment, since the noise-reduced audio signal is generated by the feedforward method, the digital filter circuit 33 is hereinafter referred to as an FF filter circuit 33.

  Just like the FB filter circuit 23, the FF filter circuit 33 includes a DSP (Digital Signal Processor) 332, an A / D conversion circuit 331 provided in the preceding stage, and a D / A conversion circuit 333 provided in the subsequent stage. Composed.

  Then, as shown in FIG. 7, the analog audio signal obtained by collecting the sound with the microphone 31 is supplied to the FF filter circuit 33 through the microphone amplifier 32 and converted into a digital audio signal by the A / D conversion circuit 331. . Then, the digital audio signal is supplied to the DSP 332.

  The DSP 332 is configured with a digital filter for generating a feedforward digital noise reduced audio signal. The digital filter generates the digital noise-reduced audio signal having a characteristic corresponding to a filter coefficient as a parameter set in the digital audio signal input thereto. In this embodiment, the filter coefficient set in the digital filter of the DSP 332 is supplied from the memory 34 through the memory controller 35.

  In this embodiment, the memory 34 is described later in order to reduce noise in a plurality of different noise environments using a noise-reduced audio signal by a feedforward method generated by a digital filter of the DSP 332. Filter coefficients as a plurality of parameters are stored.

  The memory controller 35 reads one specific (one set) filter coefficient from the memory 34 and sets it as a digital filter of the DSP 332.

  In this embodiment, the operation output signal of the operation unit 36 is supplied to the memory controller 35, and the memory controller 35 specifies from the memory 34 according to the operation output signal from the operation unit 36. One (one set) of filter coefficients is selected and read out, and set in the digital filter of the DSP 332.

  The digital filter of the DSP 332 generates a digital noise reduced audio signal corresponding to the filter coefficient that is selectively read from the memory 34 via the memory controller 35 and set.

  The digital noise reduced sound signal generated by the DSP 332 is converted into an analog noise reduced sound signal by the D / A conversion circuit 333. The analog noise reduced audio signal is supplied to the adder circuit 14 as an output signal of the FF filter circuit 33.

  An input sound signal (music signal or the like) S that the listener 1 wants to listen to through headphones is supplied to the adder circuit 14 through the sound signal input terminal 12 and the equalizer circuit 13. The equalizer circuit 13 corrects the sound characteristics of the input audio signal.

  The audio signal resulting from the addition by the adder circuit 14 is supplied to the driver 11 through the power amplifier 15 and reproduced as sound. The sound reproduced and emitted by the driver 11 includes an acoustic reproduction component by the noise-reduced sound signal generated by the FF filter 33. The sound reproduction component and the noise 3 ′ of the noise-reduced sound signal out of the emitted sound reproduced by the driver 11 are acoustically synthesized, so that the noise 3 ′ is reduced at the noise cancellation point Pc. (Cancelled)

  The memory 34, the memory controller 35, and the operation unit 36 in the second embodiment are configured in exactly the same manner as the memory 24, the memory controller 25, and the operation unit 26 in the first embodiment. Each time the button is pressed, filter coefficients corresponding to different noise environments are sequentially and cyclically changed from the memory 34 and supplied to the FF filter circuit 33.

  The configuration of the FF filter circuit 33 is exactly the same as that of the FB filter circuit 23. However, in the first embodiment and the second embodiment, the filter coefficients supplied to the digital filter configured by the DSP 232 and the DSP 332 are as follows. The first embodiment is different from the feedback method, whereas the second embodiment is different from the feed forward method.

  Next, the noise reduction operation of the feedback type noise reduction apparatus will be described with reference to FIG. 8 using a transfer function. FIG. 8 is a block diagram showing each part using its transfer function corresponding to the block diagram shown in FIG.

  In FIG. 8, A is a transfer function of the power amplifier 15, D is a transfer function of the driver 11, M is a transfer function corresponding to the microphone 31 and the microphone amplifier 32, and -α is a filter designed for feedforward. Is the transfer function. H is a transfer function of the space from the driver 11 to the cancellation point Pc, and E is a transfer function of the equalizer 13 applied to the audio signal S for listening. F is a transfer function from the position of the noise N of the external noise source 3 to the position of the listener's ear cancellation point Pc.

  When expressed as shown in FIG. 8, the block of FIG. 8 can be expressed by (Equation 5) of FIG. F ′ represents a transfer function from the noise source to the microphone position. Each of the above transfer functions is assumed to be expressed in a complex manner.

  Here, considering the ideal state, if the transfer function F can be expressed as (Equation 6) in FIG. 3, (Equation 5) in FIG. 3 can be expressed as (Equation 7) in FIG. The noise is canceled and only the music signal (or the music signal intended for listening) S remains, and it can be seen that the sound similar to the normal headphone operation can be heard. The sound pressure P at this time is expressed as (Equation 7) in FIG.

  However, in actuality, it is difficult to construct a complete filter having a transfer function such that (Equation 6) in FIG. Especially for the mid-high range, this active noise reduction processing is usually done for the mid-high range because the individual differences are great depending on the wearer and ear shape, and the characteristics change depending on the noise position and microphone position. In many cases, the headphone housing 2 performs passive sound insulation.

  Note that (Equation 6) in FIG. 3 is self-evident from the mathematical expression, but means that the transfer function from the noise source to the ear position is imitated by an electric circuit including the transfer function α of the digital filter. ing.

  Note that, as shown in FIG. 7, the cancellation point of the feed forward type of the second embodiment is set at an arbitrary ear position of the listener, unlike the feedback type of the first embodiment shown in FIG. can do.

  However, in the normal case, α is fixed, and in the design stage, some target characteristics are determined, and depending on the person, the shape of the ear is different. If it is not obtained or noise components are added in a non-reverse phase, phenomena such as abnormal noise may occur.

  In general, as shown in FIG. 9, the feedforward method of the second embodiment has low possibility of oscillation and high stability, but it is difficult to obtain a sufficient amount of attenuation. The feedback system of the embodiment requires attention to the stability of the system instead of expecting a large attenuation.

  Note that the memory controller 35 in the above-described embodiment can also be configured in the DSP 332. The equalizer circuit 13 can also be configured in the DSP 332 to convert the audio signal S into a digital signal and supply the digital signal to the equalizer circuit in the DSP 332.

[Third Embodiment and Fourth Embodiment]
In the first and second embodiments described above, the filter circuit is digitized, and a plurality of filter coefficients are prepared in a memory, and an appropriate filter is appropriately selected from the plurality of filter coefficients. The configuration is such that the coefficient can be selected and set to the digital filter.

  However, the digitalized FB filter circuit 23 and FF filter circuit 33 have a problem of delay in the A / D conversion circuits 231 and 331 and the D / A conversion circuits 233 and 333. This delay problem will be described below with respect to a feedback type noise reduction system.

  For example, as a general example, when an A / D conversion circuit and a D / A conversion circuit with a sampling frequency Fs of 48 kHz are used, the amount of delay applied inside these A / D conversion circuit and D / A conversion circuit is A Assuming 20 samples each in the / D conversion circuit and D / A conversion circuit, a total of 40 sample delays are included in the block of the FB filter circuit 23 in addition to the operation delay in the DSP, and as a result, the delay is opened. This is applied to the entire system as a loop delay.

  Specifically, the gain and phase corresponding to a delay of 40 samples at a sampling frequency of 48 kHz are shown in FIG. 10A, and the phase rotation starts from several tens of Hz until the frequency reaches Fs / 2 (24 kHz). It is rotating a lot. As shown in FIG. 11, the delay of one sample at a sampling frequency of 48 kHz is 180 deg. At a frequency of Fs / 2. This corresponds to a delay of (π), and similarly, it can be easily understood that the delay of 2 samples and 3 samples corresponds to a delay of 2π and 3π.

  On the other hand, FIG. 12 shows the measurement of the transfer function from the position of the driver 11 to the microphone 21 in the headphone configuration having an actual noise reduction system based on the feedback configuration. In this case, the microphone 21 is disposed in the vicinity of the front surface of the diaphragm of the driver 11, and it can be seen that the phase rotation is relatively small because the distance between the two is short.

  The transfer function shown in FIG. 12 corresponds to ADHM in (Equation 1) and (Equation 2), and this is multiplied by a filter having the characteristics of transfer function -β on the frequency axis, and is opened as it is. It becomes a loop. The shape of the open loop needs to satisfy the above-described conditions shown in (Equation 2) and FIG.

  Here, looking again at the phase characteristics of FIG. 10 (B), it can be seen that the rotation starts one round (2π) around 1 kHz starting from 0 deg. In addition to this, also in the ADHM characteristic of FIG. 12, there is a phase delay due to the distance from the driver 11 to the microphone 21.

  In the FB filter circuit 23, a digital filter unit configured in a DSP 232 that can be freely designed is connected in series with the delay component in the A / D conversion circuit 231 and the D / A conversion circuit 233. However, in this digital filter section, it is basically difficult to design a phase advance filter from the viewpoint of causality. However, depending on the configuration of the filter shape, there can be a “partial” phase advance only in a specific band, but it is impossible to make a wide-band phase advance circuit that compensates for the phase rotation due to this delay.

  Considering this, even if a suitable digital filter having a transfer function −β is designed by the DSP 232, in this case, the band where the noise reduction effect can be obtained by the feedback configuration is around 1 kHz where the phase rotates once. Assuming an open loop that also incorporates ADHM characteristics and considering the phase margin and gain margin, the attenuation amount and attenuation band are further narrowed.

  In this sense, a desirable β characteristic (phase inversion system in a block of transfer function −β) with respect to the characteristic as shown in FIG. 12 is a band in which the gain shape aims for a noise reduction effect as shown in FIG. It can be seen that in FIG. 13, the phase rotation does not occur so much while having a substantially mountain shape (in FIG. 13, the phase characteristic does not rotate once from the low range to the high range). Therefore, the immediate goal is to design the entire system so that the phase does not rotate once.

  Essentially, if the phase rotation is small in the noise reduction target band (mainly the low band), the phase change outside the band does not matter as long as the gain is reduced. However, in general, if there is a large amount of phase rotation in the high frequency range, this has a considerable influence on the low frequency range. Therefore, it is an object of this embodiment to design a small phase rotation for a wide band.

  In the analog circuit, the characteristics as shown in FIG. 13 can be designed, and in that sense, the noise reduction effect is greater than in the case of designing the system with the analog circuit in exchange for the merit of the digital filter described above. It is not preferable to damage.

  By the way, if the sampling frequency is increased, the delay in the A / D conversion circuit and the D / A conversion circuit can be reduced. However, a high sampling frequency is very expensive as a product and can be realized for military use or industrial use. However, as a product for general consumers such as a headphone device for listening to music, the price becomes too high and the practicality is low.

  Thus, the third and fourth embodiments provide a technique that can increase the noise reduction effect while taking advantage of the advantages of digitization in the first and second embodiments. To do.

  FIG. 14 is a block diagram illustrating a configuration of a headphone device according to the third embodiment. In the third embodiment, the configuration of the noise reduction device unit 20 using the feedback method of the first embodiment is improved.

  In the third embodiment, as shown in FIG. 14, the configuration of the FB filter circuit 23 is changed from a digital processing system including an A / D conversion circuit 231, a DSP 232, and a D / A conversion circuit 233 to an analog filter circuit 234. An analog processing system is provided in parallel.

  Then, the analog noise reduced sound signal generated by the analog filter circuit 234 is added to the adder circuit 14. The rest of the configuration is exactly the same as that shown in FIG.

  Note that the analog filter circuit 234 in FIG. 14 actually includes a case in which the input audio signal is directly passed through and supplied to the adder circuit 14 without performing filter processing on the input audio signal. . In that case, since the analog element does not exist in the analog processing system, the system is highly reliable in terms of variation and stability.

  In the FB filter circuit 23 of the third embodiment, the result of adding both after processing in parallel in the digital processing system and the analog processing system is the gain as shown in FIG. 13 as the characteristic of the transfer function β. The filter coefficients stored in the memory 24 described above are designed so as to have characteristics and phase characteristics.

  According to the third embodiment, by adding an analog processing system path in parallel to a digital processing system path, the above-described problems can be reduced and good noise reduction can be performed according to various noise environments. Can do.

  FIG. 15 shows characteristics when an analog processing path (in the case of through) is added in parallel to a digital processing path. FIG. 15A shows the leading part (up to 128 samples) of the impulse response of the transfer function in this example, FIG. 15B shows the phase characteristics, and FIG. 15C shows the gain characteristics. ing.

  From FIG. 15B, according to the third embodiment, the phase rotation is suppressed by adding an analog path, and the phase does not rotate even from the low range to the high range. I understand.

  If each characteristic is seen from another aspect, the low frequency characteristic that is the center of noise reduction is greatly influenced by the processing system by the digital filter. On the other hand, due to the delay in the A / D conversion circuit and D / A conversion circuit, For the mid-high range where the phase rotation tends to be large, the characteristics of the analog path having a quick response are effectively used.

  Thus, according to the third embodiment, it is possible to provide a noise reduction device and a headphone device capable of noise reduction adapted to various noise environments without increasing the configuration scale.

  The third embodiment is a case where noise reduction by a feedback method is performed. However, the third embodiment can be similarly applied to the case of performing noise reduction by a feedforward method according to the second embodiment.

  The fourth embodiment improves the problem in the case where only the above-described digital filter is used in the second embodiment that performs noise reduction of this feedforward method, and a configuration example thereof is shown in FIG.

  That is, in the fourth embodiment, the configuration of the FF filter circuit 33 is changed to a digital processing system including an A / D conversion circuit 331, a DSP 332, and a D / A conversion circuit 333, and an analog processing system including an analog filter circuit 334. It shall be provided in parallel.

  Then, the analog noise reduced sound signal generated by the analog filter circuit 334 is added to the adder circuit 14. The rest of the configuration is exactly the same as that shown in FIG.

  Note that the analog filter circuit 334 in FIG. 16 includes a case where the input audio signal is directly passed through and supplied to the adder circuit 14 without filtering the input audio signal. In that case, since the analog element does not exist in the analog processing system, the system is highly reliable in terms of variation and stability.

  In the FF filter circuit 33 of the fourth embodiment, the result of adding both after processing in parallel in the digital processing system and the analog processing system is the gain as shown in FIG. 13 as the characteristic of the transfer function α. The filter coefficients stored in the memory 34 described above are designed so as to have characteristics and phase characteristics.

  Note that the memory controllers 25 and 35 in the above-described embodiment may be configured in the DSPs 232 and 332. The equalizer circuit 13 can also be configured in the DSPs 232 and 332 to convert the audio signal S into a digital signal and supply it to the equalizer circuit in the DSPs 232 and 332.

[Fifth Embodiment]
As described above, the feedforward method of the second embodiment has low possibility of oscillation and high stability, but it is difficult to obtain a sufficient attenuation amount, while the feedback method of the first embodiment is difficult. However, instead of expecting a large amount of attenuation, attention must be paid to the stability of the system.

  Therefore, in the fifth embodiment, a noise reduction method having both advantages is provided. That is, in the fifth embodiment, as shown in FIG. 17, both the feedback type noise reduction device unit 20 and the feedforward type noise reduction device unit 30 are provided.

  In FIG. 17, the block configuration is shown using a transfer function. In the feedback-type noise reduction device unit 20, the transfer function corresponding to the microphone 21 and the microphone amplifier 22 is generated by the M1 and FB filter circuit 23. A1 is a transfer function of a power amplifier that outputs and amplifies the noise-reduced audio signal, and D1 is a transfer function of a driver that reproduces the noise-reduced audio signal. The spatial transfer function from the driver to the cancellation point Pc is H1.

  Further, in the noise reduction device unit 30 of the feedforward method, the transfer function corresponding to the microphone 31 and the microphone amplifier 32 is M2, and the transfer function of the power amplifier that outputs and amplifies the noise-reduced audio signal generated by the FB filter circuit 33. Is A2, and the transfer function of the driver that reproduces the noise-reduced audio signal is D2. The spatial transfer function from the driver to the cancellation point Pc is H2.

  In the embodiment of FIG. 17, the memory 34 stores a plurality of sets of filter coefficients to be supplied to the FB filter circuit 23 and the FF filter circuit 33, respectively. An appropriate filter coefficient is selected from a plurality of sets of filter coefficients for each in accordance with the user's button operation through the operation unit 36 as described above, and set to the respective filter circuits 23 and 33. It is configured.

  In the example of FIG. 17, a system that acoustically reproduces the noise-reduced audio signal generated by the feedback-type noise reduction apparatus unit, and a system that acoustically reproduces the noise-reduced audio signal generated by the feed-forward type noise reduction apparatus unit, Are provided separately. In the example of FIG. 17, the power amplifier and driver of the system that reproduces the noise-reduced audio signal generated by the feedback-type noise reduction device unit is used only for noise reduction, and the feed-forward type noise reduction device unit. The power amplifier and driver of the system that acoustically reproduces the noise-reduced audio signal generated in (1) are used not only for noise reduction but also for acoustic reproduction of the audio signal S to be listened to.

  Further, in the example of FIG. 17, the audio signal S to be listened to is converted into a digital audio signal by the A / D conversion circuit 37 and then supplied to the DSP 332 of the FF filter circuit 33. Although not shown, the DSP 332 in this example includes not only a digital filter for generating a feedforward noise reduction audio signal, but also an equalizer circuit for adjusting the audio characteristics of the audio signal S to be listened to; An adder circuit is configured, and the output audio signal of the equalizer circuit and the noise-reduced audio signal generated by the digital filter are added by the adder circuit and output from the DSP 332.

  In the fifth embodiment, the feedback-type noise reduction unit 20 and the feed-forward type noise reduction unit 30 perform the above-described noise reduction processing operation independently of each other. However, the noise cancellation point Pc is set to the same position in both systems.

  Therefore, according to the fifth embodiment, it is possible to realize a noise reduction system in which the feedback type and feedforward type noise reduction processes operate in a complementary manner and the advantages of both types can be obtained.

  In FIG. 17, the filter coefficient of the digital filter is changed in both the feedback method and the feedforward method. However, the filter coefficient is changed only for the digital filter of one method, for example, only the digital filter of the feedforward method. You may comprise so that selection change is possible.

  In the example of FIG. 17, the FB filter circuit 23 and the FF filter circuit 33 are configured as separate DSPs, but the entire circuit configuration is simplified by configuring as one DSP. be able to. In the example of FIG. 17, the power amplifier and the driver are also provided separately for the feedback type noise reduction device unit 20 and the feedforward type noise reduction device unit 30, but the same as in the above-described embodiment. In addition, the power amplifier 15 and the driver 11 may be used. An example of such a configuration is shown in FIG.

  That is, in the example of FIG. 18, a filter circuit 40 including an A / D conversion circuit 41, a DSP 42, and an A / D conversion circuit 43 is provided. The analog audio signal from the microphone amplifier 21 is converted into a digital audio signal by the A / D conversion circuit 44 and supplied to the DSP 42. Further, the audio signal S to be listened input through the input terminal 12 is converted into a digital audio signal by the A / D conversion circuit 37 and supplied to the DSP 42.

  In this example, as shown in FIG. 19, the DSP 42 includes a digital filter circuit 421 for obtaining a feedback-type noise-reduced audio signal, and a digital filter circuit 422 for obtaining a feed-forward-type noise-reduced audio signal. The digital equalizer circuit 423 and the adding circuit 424 are configured.

  Then, the digital audio signal from the A / D conversion circuit 44 (the digital signal of the sound collected by the microphone 21) is supplied to the digital filter circuit 421, and the digital audio signal from the A / D conversion circuit 41 (by the microphone 31). The collected digital audio signal is supplied to the digital filter circuit 422, and the digital audio signal from the A / D conversion circuit 37 (listening target audio digital signal) is supplied to the equalizer circuit 423.

  Further, as described above, in this example, the memory 34 includes a plurality (multiple sets) of filter coefficients for the digital filter circuit 421 and a plurality (multiple sets) of filter coefficients for the digital filter circuit 422. The memory controller 35 selects filter coefficients for the digital filter circuit 421 and the digital filter 422 from the memory 34 in response to a user operation through the operation unit 36, and the digital filter circuit 421 and The digital filter circuit 422 is supplied.

  The memory 34 also stores parameters that make the equalizer characteristics of the digital equalizer circuit 423 correspond to a plurality (a plurality of sets) of filter coefficients for the digital filter 422. The memory controller 35 includes an operation unit In response to a user operation through 36, an equalizer characteristic parameter is selectively read out from the memory 34 in accordance with selection of a filter coefficient for the digital filter circuit 422 and supplied to the digital equalizer circuit 423.

  The noise-reduced audio signal generated by the digital filter circuit 421 and the digital filter circuit 422 and the digital audio signal from the equalizer circuit 423 are supplied to the addition circuit 424 and added, and the addition result is a D / A conversion circuit. 43 is converted into an analog audio signal. The analog audio signal from the D / A conversion circuit 43 is supplied to the driver 11 through the power amplifier 15. As a result, the noise 3 ′ is reduced (cancelled) at the noise cancellation point Pc.

  In FIG. 18, reference numerals 40a, 40b, 40c, and 40d denote connections in which connection cables are connected between the noise reduction device unit and the driver 11, microphone 21, microphone 31, input end 12 (headphone plug), and the like. It is a terminal part.

[Sixth Embodiment]
In the sixth embodiment, similar to the third and fourth embodiments described above, the fifth embodiment is only digital processing, and the delay of the A / D conversion circuit and the D / A conversion circuit is reduced. Considering that there is a problem, this is an embodiment in which the problem is improved.

  That is, in the sixth embodiment, an analog filter system is provided in parallel with the digital filter system, as in the third and fourth embodiments shown in FIGS. FIG. 20 shows a block diagram of an example of the noise reduction device unit 50 in the case of the sixth embodiment.

  In the noise reduction device section 50 of the sixth embodiment, as shown in FIG. 20, an analog filter circuit 51 for generating a feedback type analog noise reduced voice signal, and a feed forward type analog noise reduced voice signal are provided. 19 is added to the configuration of FIG. 19.

  The analog audio signal from the microphone amplifier 22 is supplied to the A / D conversion circuit 44 and also supplied to the analog filter circuit 51 for generating a feedback type analog noise reduction audio signal. The analog noise reduced audio signal from the analog filter circuit 51 is supplied to the adder circuit 53.

  The analog audio signal from the microphone amplifier 32 is supplied to the A / D conversion circuit 41 and also supplied to the analog filter circuit 52 for generating a feedforward analog noise reduction audio signal. The analog noise reduced audio signal from the analog filter circuit 52 is supplied to the adder circuit 53.

  In addition, the addition circuit 53 is further supplied with an addition signal of the noise-reduced audio signal and the listening target audio signal from the filter circuit 40. Then, the audio signal from the adder circuit 53 is supplied to the driver 11 through the power amplifier 15. Thereby, in this embodiment, the noise reduction process of the feedback method and the noise reduction process of the feedforward method are used in combination, and the problem in the case of generating the noise-reduced sound signal only by the digital filter is solved. It is possible to provide a noise reduction device and a headphone device that can be realized for consumers.

[Modification of Manual Selection Method (First to Sixth Embodiments)]
In the first to sixth embodiments described above, every time the push switch of the operation unit 26 is pressed, the filter coefficients corresponding to different noise environments are sequentially and cyclically changed from the memory 24 to thereby change the FB filter. Each time the listener pushes the push switch, the name of each noise environment (“station platform”, “airfield”, “in the train”, etc.) is displayed on the display unit. In addition, the addition unit 14 adds the audio signal having the name of each noise environment to the audio signal to be acoustically reproduced by the driver 11 so as to notify the user of which noise environment the filter coefficient is changed to. Also good.

  If the noise reduction device has a display screen, a list of noise environment names corresponding to each of a plurality of selectable filter coefficients is displayed on the display screen. It is also possible to select and specify the filter coefficient of the noise environment considered as follows.

  The operation units 26 and 36 are not limited to push switches, and operation means having various configurations can be used. For example, when the listener 1 is tapped (tapped) on the headphone housing 2 by using a vibration sensor or the like, the detection output is the following filter coefficient in the same manner as when the push switch is pressed. The change timing may be used.

  In the above-described embodiment, the filter coefficient is changed every time a user operation is performed. However, if there is a user operation, the memory controller 25 or 35 stores a plurality of filter coefficients from the memory 24 or 35. One by one may be set in the digital filter sequentially for a predetermined period, and the listener may listen to the predetermined period.

  In that case, after finishing listening to all the filter coefficients, the user decides which number of filter coefficients from the listener should receive the input of optimization, or the optimum filter coefficient. When the filter coefficient is being selected, the user performs a predetermined user operation so that the user determines the optimum filter coefficient. In the latter case, the operation of selecting a plurality of filter coefficients in order and allowing the listener to listen for a predetermined time may be repeated several times for the plurality of filter coefficients.

  When the user determines whether or not the filter coefficient is in an optimal state, if the sound signal S to be listened to is reproduced and the determination is difficult, the user has changed the filter coefficient. The audio signal S may be forcibly muted for a predetermined time so that the user can determine the noise reduction effect.

[Automatic change method]
The above first to sixth embodiments are all cases where the filter coefficient set in the digital filter is selected and set in accordance with the user's operation. However, the embodiments described below are automatically In addition, the filter coefficient is set according to the noise environment where the headphone device is used.

  As described below, there are several examples of the configuration for automatically setting the filter coefficient according to the noise environment of the place where the headphone device is used as described below. By applying each example instead of manual selection based on the operation of the operation units 26 and 36 in the first to sixth embodiments described above, the noise of the configuration of the first to sixth embodiments, respectively. Applicable to a reduction device. Hereinafter, some embodiments will be described.

[Seventh Embodiment]
The seventh embodiment employs an automatic selection method as described below in place of the operation unit 26 in the configuration of the third embodiment, which is the feedback method described above and has an analog filter system in parallel. Embodiment. FIG. 21 is a block diagram showing a configuration example of the headphone device according to the seventh embodiment.

  In the seventh embodiment, the DSP 232 of the FB filter circuit 23 includes not only a digital filter circuit 2321 corresponding to the feedback method but also a noise analysis unit 2322 and an optimum filter coefficient evaluation unit 2323.

  The noise analysis unit 2322 analyzes the characteristics of the noise collected by the microphone 21 and supplies the analysis result to the optimum filter coefficient evaluation unit 2323. In this embodiment, the optimum filter coefficient evaluation unit 2323 stores, in the memory 24, a filter coefficient having a noise reduction curve characteristic closest to a noise waveform curve and a reverse characteristic curve based on the analysis result from the noise analysis unit 2322. The filter coefficient is selected from a plurality of filter coefficients, and an optimum filter coefficient (one set) is determined, and the determination result is supplied to the memory controller 25.

  The memory controller 25 receives the optimum filter coefficient determination result from the optimum filter coefficient evaluation unit 2323, reads out the filter coefficient corresponding to the optimum filter coefficient determination result from the memory 24, and supplies the filter coefficient to the digital filter circuit 2321. To set.

  In the seventh embodiment, the above-described automatic filter coefficient automatic selection processing operation is configured to be activated by an activation control signal from the activation control unit 61. That is, the activation control signal from the activation control unit 61 is supplied to the memory controller 25 and also to the noise analysis unit 2322 and the optimum filter coefficient evaluation unit 2323.

  Further, since it is better to perform noise analysis in an environment where there is no sound reproduction sound due to the audio signal S to be listened to, in the seventh embodiment, the audio signal S input through the input terminal 12 is the equalizer circuit 13. And is also supplied to the activation control unit 61. A muting circuit 16 that mutes the audio signal S is provided between the equalizer circuit 13 and the adder circuit 14.

  The activation control unit 61 determines the presence or absence of the audio signal S when attempting to activate the automatic filter coefficient automatic selection processing operation, and determines that the audio signal S is present when the audio signal S is present. In the circuit 16, the audio signal S from the equalizer circuit 13 is muted for a predetermined time, and the sound is picked up by the microphone 21 so that there is no reproduced sound by the audio signal S. The predetermined time in this case is a time necessary for performing noise analysis so that the optimum filter coefficient can be selected.

In this embodiment, the activation control unit 61 activates the optimum filter coefficient automatic selection processing operation at the following timing. In other words, the startup timing is
(1) When the power is turned on (2) When the listener operates the automatic selection process start switch (3) Every fixed time (4) When there is a large change in noise (5) Noise above a predetermined level is detected Such as when.

  Whether or not the power is turned on in (1) above, when the headphone device is supplied with the power supply voltage from the playback device for the audio signal S, the headphone plug constituting the input terminal 12 is inserted into the headphone jack of the playback device. Thus, the activation control unit 61 can determine whether or not the supply of the power supply voltage has been received.

  In the case of (2) above, the activation control unit 61 includes an automatic selection process activation switch (not shown), and determines whether it is the activation timing based on whether or not the automatic selection process activation switch is operated. Like that.

  Further, without providing the automatic selection processing activation switch, for example, when the listener 1 is tapped (tapped) on the headphone housing 2 is detected from the collected sound signal of the microphone 21 or 31, and the detected output May be set as the start timing of the automatic filter coefficient automatic selection processing operation.

  In the case of (3) above, the activation control unit 61 includes an interval timer (not shown), and every time a predetermined time is measured with the interval timer, the automatic filter coefficient automatic selection processing operation is performed. Try to start. In this case, the listener can set the predetermined time measured by the interval timer. For example, when the listener is moving while listening to the audio signal S from the playback device with a headphone device, the listener sets a predetermined time measured by the interval timer, and when not moving, the listener measures a predetermined time measured by the interval timer. Can be set to a long time.

  In the case of (4) above, in this embodiment, when the audio signal S is not reproduced, the activation control unit 61 collects noise at an interrupt timing of a predetermined cycle. Further, when the audio signal S is being reproduced, noise is collected in the silent section of the audio signal S. When it is determined that the difference between the collected noise and the noise collected at the previous timing is larger than a predetermined threshold value, the automatic filter coefficient automatic selection processing operation is activated. To. This is because it can be determined that the noise environment has changed when the noise changes greatly.

  In the case of (5), as in the case of (4) above, the activation control unit 61 picks up noise at an interrupt timing of a predetermined period when the audio signal S is not reproduced, When the audio signal S is being reproduced, noise is collected in the silent section of the audio signal S. Then, when it is determined that the collected noise is larger than a predetermined threshold value, the automatic filter coefficient automatic selection processing operation is activated. This is because it is considered that it is better to reduce the noise when the noise becomes loud from the low noise state.

  The above (1) to (5) are examples of the start timing of the optimum filter coefficient automatic selection processing operation, and needless to say, other timings may be used. Further, it is not necessary to use all the activation timings (1) to (5) above, and one or more activation timings among them may be used.

FIG. 22 is a flowchart showing an example of the flow of processing operations in the activation control unit 61.
That is, the activation control unit 61 monitors whether or not the activation timing of the optimum filter coefficient automatic selection processing operation has come (step S11).

  When it is determined in step S11 that the activation timing has come, the activation control unit 61 determines whether or not the audio signal S to be listened is being reproduced based on the presence or absence of the audio signal S (step S12).

  When it is determined in step S12 that the audio signal S is not being reproduced, the activation control unit 61 sends an activation control signal to the noise analysis unit 2322, the optimum filter coefficient evaluation unit 2323, and the memory controller 25 to automatically determine the optimum filter coefficient. The selection processing operation is activated (step S14).

  Also, when it is determined in step S12 that the audio signal S is being reproduced, the activation control unit 61 supplies the muting control signal to the muting circuit 16 to force the audio signal S being reproduced, Muting control is performed for a predetermined time (step S13).

  Then, the process proceeds to step S14 after step S13, and the activation control unit 61 sends an activation control signal to the noise analysis unit 2322, the optimum filter coefficient evaluation unit 2323, and the memory controller 25, and performs the automatic filter coefficient automatic selection processing operation. Try to start.

  Next, specific examples of the noise analysis unit 2322 and the optimum filter coefficient evaluation unit 2323 will be described. FIG. 23 is a first example of the configuration of a specific example of the noise analysis unit 2322 and the optimum filter coefficient evaluation unit 2323. In this example, a noise waveform is detected by noise analysis using FFT (Fast Fourier Transform) processing.

  As shown in FIG. 23, the signal from the A / D conversion circuit 231 (which consists of noise since the audio signal S does not exist when the activation is applied as described above) is sent to the low-pass filter 71 of the noise analysis unit 2322. After being supplied and the high frequency component is removed, it is supplied to the data thinning processing unit 72 and data is thinned appropriately. Then, the data from the data thinning-out processing unit 72, which is data for a predetermined period, is supplied to the FFT processing unit 73, subjected to the FFT calculation, and the FFT calculation result is supplied to the optimum filter coefficient evaluation unit 2323.

  The optimum filter coefficient evaluation unit 2323 recognizes a noise waveform curve from the FFT calculation result. Then, a filter coefficient having an attenuation curve characteristic close to the inverse of the noise waveform curve is selected from a plurality of memories 24.

  For example, when the noise reduction characteristic by the plurality of filter coefficients stored in the memory 24 is as shown in FIG. 6 described above, the noise waveform curve of the FFT calculation result has mainly energy in the low range. In such a case, (1) the filter coefficient that obtains the noise reduction characteristic of the low-frequency emphasis curve is selected as the optimum filter coefficient.

  In FIG. 23, the low-pass filter 71 and the data thinning-out processing unit 72 are used because the noise characteristic has many low-frequency components in the first place. This is because it is difficult in the first place to target the high range, and it is possible to reduce the amount of calculation by downsampling.

  In the case of this example, the FFT result for the inverse characteristic curve of the attenuation curve for each filter coefficient is stored in the memory 24, and the FFT result from the FFT processing unit 73 and each stored filter coefficient are stored. The filter coefficient corresponding to the inverse characteristic curve with a small error may be determined as the optimum filter coefficient by comparing with the FFT result for the inverse characteristic curve of the attenuation curve at this time.

  Next, a second example of specific examples of the noise analysis unit 2322 and the optimum filter coefficient evaluation unit 2323 will be described. FIG. 24 shows a second example of the configuration of a specific example of the noise analysis unit 2322 and the optimum filter coefficient evaluation unit 2323.

  In this second example, as shown in FIG. 24, the noise analysis unit 2322 includes a plurality of, in this example, six bandpass filters 81, 82, 83, 84, 85, 86, Six energy value calculation storage units 91, 92, 93, 94, 95 for calculating the energy values of the outputs of the bandpass filters 81, 82, 83, 84, 85, 86 as dB values and storing them in the built-in registers. , 96.

  In this example, the pass center frequencies of the six band pass filters 81, 82, 83, 84, 85, 86 are 50 Hz, 100 Hz, 200 Hz, 400 Hz, 800 Hz, and 1.6 kHz.

  Then, the signal from the A / D conversion circuit 231 (consisting of noise since the audio signal S does not exist when the activation is applied as described above) is generated by the six band-pass filters 81, 82, 83, 84, 85 and 86, respectively. The outputs of the six band pass filters 81, 82, 83, 84, 85, 86 are supplied to the energy value calculation storage units 91, 92, 93, 94, 95, 96, and the energy values A (0), A (1), A (2), A (3), A (4), A (5) are calculated and stored in the built-in registers.

  On the other hand, in the second example, the memory 24 corresponds to the above-described four types of noise reduction curves (1), (2), (3), and (4) as shown in FIG. 4 sets of filter coefficients are stored, and attenuation at 50Hz, 100Hz, 200Hz, 400Hz, 800Hz, 1.6kHz in each noise reduction curve (1), (2), (3), (4) A representative value (dB value) is stored corresponding to each filter coefficient.

  For example, the attenuation representative values (dB values) at 50 Hz, 100 Hz, 200 Hz, 400 Hz, 800 Hz, and 1.6 kHz in the low-frequency emphasis curve (1) are B1 (0), B1 (1), B1 (2). .. Stored in association with the corresponding filter coefficient as B1 (5), and representative attenuation values (dB values) at 50 Hz, 100 Hz, 200 Hz, 400 Hz, 800 Hz, and 1.6 kHz in the low-mid range emphasis curve (2) ) Are stored in association with the corresponding filter coefficients as B2 (0), B2 (1), B2 (2)... B2 (5).

  Then, the optimum filter coefficient evaluation unit 2323 of the second example stores the energy values A (0), A (1), A (2), A (3) stored in the energy calculation storage units 91 to 96, respectively. ), A (4), A (5) and the attenuation value representative value by the noise reduction curve by each filter coefficient stored in the memory 24 are detected, and the noise reduction curve having the smallest sum of the differences is supported. The filter coefficient to be determined is determined as the optimum filter coefficient.

  That is, the energy values A (0), A (1), A (2), A (3), A (4), A (5) and the noise reduction curve based on each filter coefficient stored in the memory 24. This is because the sum of the difference from the attenuation representative value is equal to the residual of the attenuation result by each noise reduction curve with respect to the input noise, and a smaller value means that the noise is reduced.

  An example of the flow of processing operations in the noise analysis unit 2322 and the optimum filter coefficient evaluation unit 2323 in the case of the second example is shown in the flowchart of FIG.

  First, the energy values A (0), A (1), A (2), A (3), A (4), A (5) of the outputs of the noise analysis unit 2322 bandpass filters 81-86 are calculated. Store in the register (step S21).

  Next, the optimum filter coefficient evaluation unit 2323 reads the stored energy values A (0) to A (5), performs energy → amplitude conversion, and corrects the values (step S22). In this correction, when the total selectivity Q of each of the BPFs 81 to 86 is constant, for example, when white noise having a constant frequency / amplitude value is passed, the energy value of the waveform that has passed is not constant, and the low range is greatly output. Therefore, this correction calculation is necessary. Further, correction may be necessary depending on how to select the total selectivity Q, and these are corrected together.

  Next, the optimum filter coefficient evaluation unit 2323 first obtains the representative values B1 (0) to B1 (5) of the low-frequency emphasis curve of the attenuation curve (1) from the memory 24 and the energy values A (0) to A (5). ) Are subtracted from the respective correction values (step S23).

  Next, the optimum filter coefficient evaluation unit 2323 corrects the subtraction value with the auditory characteristic curve to obtain values C1 (0) to C1 (5) (step S24). Next, the optimum filter coefficient evaluation unit 2323 calculates a total value obtained by converting the values C1 (0) to C1 (5) into linear values (step S25). This total value is an evaluation score for one attenuation curve.

  Here, the audible characteristic curve may be a so-called A curve or C curve, or may be a value obtained by converting the loudness in consideration of the absolute sound volume, or may be set uniquely.

  Then, the optimum filter coefficient evaluation unit 2323 performs the operations in steps S23 to S25 for all of the attenuation curves (1) to (4), and obtains an evaluation score corresponding to each attenuation curve (step S26). .

  Then, when the optimal filter coefficient evaluation unit 2323 can calculate the score values corresponding to all the curves, it determines that the attenuation curve having the smallest evaluation score value can expect the most noise attenuation effect, A filter coefficient corresponding to this attenuation curve is determined as an optimum filter (step S27).

  Note that the memory controller 25 in the above-described embodiment can also be configured in the DSP 232. The equalizer circuit 13 can also be configured in the DSP 232 to convert the audio signal S into a digital signal and supply the digital signal to the equalizer circuit in the DSP 232.

[Eighth Embodiment]
The eighth embodiment employs the automatic selection method described below in place of the operation unit 26 in the configuration of the fourth embodiment, which is the feedforward method described above and has an analog filter system in parallel. This is an embodiment. FIG. 27 is a block diagram showing a configuration example of the headphone device according to the eighth embodiment.

  In the eighth embodiment, the DSP 332 of the FF filter circuit 33 includes not only the digital filter circuit 3321 compatible with the feedforward method but also the noise analysis unit 3322 and the optimum filter coefficient evaluation unit, as in the seventh embodiment. 3323 is configured.

  In the eighth embodiment, the noise analysis unit 3322 analyzes the characteristics of the noise collected by the microphone 31 and supplies the analysis result to the optimum filter coefficient evaluation unit 3323. The configurations and processing operations of the noise analysis unit 3322 and the optimum filter coefficient evaluation unit 3323 are the same as those of the seventh embodiment. However, the eighth embodiment relates to the activation control of the automatic selection processing operation of the optimum filter coefficient. This is different from the seventh embodiment.

  In the above seventh embodiment, the muting is forcibly performed when the audio signal S is being reproduced. However, in the eighth embodiment, the muting of the audio signal S is not performed. The silent section is detected, and the optimum filter coefficient automatic selection processing operation is executed in the silent section.

  That is, in the eighth embodiment, the activation control unit 62 is provided, but the muting circuit 16 is not provided between the equalizer circuit 13 and the adder circuit 14. The activation control unit 62 supplies the activation control signal to the noise analysis unit 3322, the optimum filter coefficient evaluation unit 3323, and the memory controller 35.

  The memory 34 stores a plurality (multiple sets) of filter coefficients corresponding to the feedforward method as described above. Then, the memory controller 35 reads out the optimum filter coefficient from the plurality of filter coefficients in the memory 35 while receiving the activation control from the activation control unit 62, and sends it to the digital filter circuit 3321. To set. The other points are configured in the same manner as in the seventh embodiment.

  An example of the flow of the start control operation by the start control unit 62 of the eighth embodiment will be described with reference to the flowchart of FIG.

  That is, first, the activation control unit 62 monitors whether or not it is the activation timing of the automatic filter coefficient automatic selection processing operation (step S31). Also in the eighth embodiment, with respect to the activation timing, the activation timings (1) to (5) described above can be used as in the seventh embodiment.

  When it is determined in step S31 that the activation timing has come, the activation control unit 62 determines whether or not the audio signal S to be listened to is being reproduced based on the presence or absence of the audio signal S (step S32).

  When it is determined in step S32 that the audio signal S is not being reproduced, the activation control unit 62 sends an activation control signal to the noise analysis unit 3322, the optimum filter coefficient evaluation unit 3323, and the memory controller 35, and automatic optimization of the optimum filter coefficient is performed. The selection processing operation is activated (step S34).

  If it is determined in step S32 that the audio signal S is being reproduced, the activation control unit 62 monitors the silent period of the audio signal S to determine whether or not the silent signal has been reached (step S33). When the silent section is determined, the process proceeds to step S34, and the activation control unit 62 sends an activation control signal to the noise analysis unit 2322, the optimum filter coefficient evaluation unit 2323, and the memory controller 35, and activates the optimum filter coefficient automatic selection processing operation. To apply.

  Since the optimum filter coefficient automatic selection processing operation is the same as that of the seventh embodiment in the eighth embodiment, the description thereof is omitted.

  Note that the memory controller 35 in the above-described embodiment can also be configured in the DSP 332. The equalizer circuit 13 can also be configured in the DSP 332 to convert the audio signal S into a digital signal and supply the digital signal to the equalizer circuit in the DSP 332.

[Ninth Embodiment]
In the seventh embodiment and the eighth embodiment described above, the optimum filter coefficient automatic selection processing operation is the start timing and forcibly cut off the reproduced audio signal to generate a silent section, In the ninth embodiment, noise is generated by removing the component of the reproduced sound signal S from the sound signal picked up from the microphone 31. Only, and noise analysis is performed on the extracted noise. Thereby, it is possible to perform noise measurement while accurately playing reproduced sound.

  A case where the configuration example of the headphone device of the ninth embodiment is applied to a feedforward noise reduction device will be described. FIG. 29 is a block diagram illustrating a configuration example of the headphone device in that case.

  As shown in FIG. 29, the transfer function from the driver 11 inside the headphone housing 2 to the microphone 31 outside the headphone housing 2 is now H. This transfer function H can be made known by measuring it in advance.

  This transfer function H itself includes many resonances and reflections in the headphone housing 2 and is often complicated. Actually, it is assumed that a transfer function H ′ approximating the feature of H is used due to the amount of calculation. In many cases, when calculating using the transfer function H, the impulse response h is often FIR (Finite Impulse Response) calculation. However, since FIR calculation consumes a lot of computer resources, DSP calculation consumes a lot of computer resources. The feature of the transfer function H is approximated as a transfer function H ′, and this is realized as an IIR (Infinite Impulse Response) filter.

  In the ninth embodiment, as shown in FIG. 29, the DSP 332 includes a digital filter circuit 3321, a noise analysis evaluation unit 3324 including the noise analysis unit 3322 and the optimum filter coefficient evaluation unit 3323 described above, and a digital equalizer circuit. 3325, a transfer function H ′ multiplication unit 3326, a subtraction circuit 3327, and an addition circuit 3328 are configured.

  In the example of FIG. 29, the audio signal S through the input terminal 12 is converted into a digital audio signal by the A / D conversion circuit 37 and supplied to the equalizer circuit 3325 of the DSP 332 of the FF filter circuit 33.

  The output signal of the equalizer circuit 3325 is supplied to the D / A conversion circuit 333 through the adder circuit 3328 and also supplied to the transfer function H ′ multiplier 3326. The transfer function H ′ multiplier 3326 multiplies the output signal of the equalizer circuit 3325 by the transfer function H ′ and supplies the result to the subtraction circuit 3327.

  The subtracting circuit 3327 is supplied with a reproduction acoustic signal of the audio signal S including the noise 3 picked up by the microphone 31 from the A / D conversion circuit 331 through the microphone amplifier 32, and the audio signal S including the noise 3 is supplied. Is subtracted from the audio signal from the transfer function H ′ multiplier 3326.

  Since the transfer function H ′ is a transfer function from the driver 11 inside the headphone housing 2 to the microphone 31 outside the headphone housing 2, the sound signal from the transfer function H ′ multiplier 3326 is collected by the microphone 31. This corresponds to the reproduced sound signal of the sound signal S to be played. Therefore, only the noise 3 component is obtained from the subtraction circuit 3327. An output signal of the subtraction circuit 3327 is supplied to the noise analysis evaluation unit 3324.

  In the noise analysis / evaluation unit 3324, as described above, the noise component that is the input signal is analyzed by the noise analysis unit, and the noise analysis result is supplied to the optimum filter coefficient evaluation unit. Then, the optimum filter coefficient evaluation unit determines the optimum filter coefficient as described above, and supplies the determination result to the memory controller 35. The memory controller 35 reads out the optimum filter coefficient from the memory 34 based on the determination result of the optimum filter coefficient and sets it in the digital filter circuit 3321.

  The noise-reduced audio signal generated by the digital filter circuit 3321 is supplied to the adder circuit 3328 and added to the audio signal from the equalizer circuit 3325. Then, the added output signal is supplied to the D / A conversion circuit 333.

  As described above, in the ninth embodiment, the time waveform estimated at the sound collection point of the microphone 31 is obtained by adopting the configuration as shown in FIG. Thus, a difference can be obtained from the collected sound signal from the microphone 31, and only the actual noise component can be extracted without cutting the reproduced sound of the sound signal S.

[Other Embodiments and Modifications of Automatic Selection Method]
In the seventh to ninth embodiments described above, the noise collected by the microphone 21 or 31 is analyzed, and the optimal filter coefficient is selected using the analysis result. The filter coefficient can be automatically selected.

  That is, in the feedback type noise reduction device, the sound at the noise cancellation point Pc is collected by the microphone 21, so it is confirmed whether the noise is reduced (cancelled) from the sound signal of the sound collected by the microphone 21. can do.

  Therefore, in the feedback type noise reduction device, when the start timing comes, the memory controller 25 or 35 sequentially selects each of the plurality of filter coefficients from the memory 24 or 35 for a predetermined period of time. The digital filter is set, and the residual noise at the noise cancellation point Pc for each filter coefficient is collected by the microphone 21 and evaluated. Then, the filter coefficient with the smallest residual noise is determined as the optimum filter coefficient.

  Also in this case, when the evaluation is performed, the audio signal S is muted, or a silent section of the audio signal S is detected to remove the influence of the audio signal S. Similarly to the embodiment of FIG. 9, the audio signal S is multiplied by the transfer function H ′ and subtracted from the audio signal from the microphone 21, and residual noise is detected and evaluated for the subtracted output. Also good.

  In the case of the feedforward method, by providing a microphone that picks up the sound of the noise cancellation point Pc, the noise cancellation point Pc residual noise is evaluated and the optimum filter coefficient is automatically determined as described above. can do.

  In the case of the combined method of the feedforward method and the feedback method, it is possible to automatically determine the optimum filter coefficient by evaluating the noise cancellation point Pc residual noise with a microphone that picks up the sound of the noise cancellation point Pc. Needless to say.

[Other Embodiments and Modifications]
In the description of each of the above-described embodiments, in the FB filter circuit and the FF filter circuit, the digital filter circuit is configured using a DSP, but a software program is used using a microcomputer (or a microprocessor) instead of the DSP. Thus, the digital filter circuit can be processed.

  When a microcomputer (or a microprocessor) is used instead of the DSP, the memory controller portion can also be configured by the software program. Conversely, it is also possible to configure the memory controller portion in the DSP.

  In the first to fourth embodiments and the seventh and eighth embodiments described above, the equalizer circuit 13 is configured as an analog circuit, but the fifth, sixth, and ninth embodiments. Similarly to the above, the digital equalizer circuit may be configured in the DSP, or may be configured by a microcomputer software program.

  In the case of a device that uses a microphone 21 and a microphone 31, as in the fifth embodiment shown in FIG. May use either the microphone 21 or the microphone 31 or both.

  In the seventh to eighth embodiments, the noise analysis is performed and the optimum filter coefficient is selected. However, if the noise analysis can be performed accurately, the noise analysis result is used. It is expected that a filter coefficient capable of estimating an attenuation curve and obtaining the estimated attenuation curve can be calculated. By doing so, it is not necessary to store a plurality of filter coefficients in the memory.

  However, noise analysis for estimating such an attenuation curve may require a fine FFT or use a large amount of bandpass filter, which may make the configuration complicated and expensive. There is. In that respect, in the above-described embodiment, an accurate attenuation curve is not required, and it is only necessary to determine which attenuation curve is optimal among the attenuation curves based on a plurality of filter coefficients prepared in advance. Therefore, it can be configured easily and inexpensively.

  Moreover, although the above embodiment demonstrated the case where the noise reduction audio | voice output apparatus of embodiment of this invention was a headphone apparatus, communication terminals, such as an earphone apparatus or headset apparatus provided with a microphone, and also a mobile telephone terminal It can also be applied to. In addition, the noise reduction audio output device according to the embodiment of the present invention can be applied to a portable music playback device combined with headphones, earphones, and a headset.

  In addition, in the above-described embodiment, the noise reduction device unit is provided on the headphone device side. It is also possible to provide a noise reduction device unit on the playback device side.

  DESCRIPTION OF SYMBOLS 1 ... Listener, 2 ... Headphone housing, 3 ... Noise source, 11 ... Headphone driver, 12 ... Audio signal input terminal, 13 ... Equalizer circuit, 14 ... Adder circuit, 15 ... Power amplifier, 21, 31 ... Microphone, 23 ... FB filter circuit, 33 ... FF filter circuit, 24, 34 ... Memory, 25, 35 ... Memory controller, 26, 36 ... Operation unit, 231, 331 ... A / D conversion circuit, 232, 332 ... DSP, 233, 333 ... D / A conversion circuit, 2322, 3322 ... noise analysis unit, 2323, 3323 ... optimum filter coefficient evaluation unit, 61, 62 ... start-up control unit

Claims (8)

  1. A microphone that is installed toward the outside of the headphone housing and collects external noise from the headphone;
    A / D conversion means for converting an analog audio signal picked up by the microphone into a digital audio signal;
    Muting means for muting a predetermined input audio signal;
    When the predetermined input audio signal is muted by the muting means, the digital audio signal obtained by the A / D conversion means is subjected to high-frequency component removal and downsampling, and noise is reduced . Noise characteristic analysis means for analyzing characteristics;
    Holding means for holding a plurality of sets of parameters prepared in advance according to a plurality of types of noise characteristics;
    A selection setting means for selecting a parameter set suitable for noise reduction, which is a noise characteristic of an analysis result by the noise characteristic analysis means, from among a plurality of sets of parameters held in the holding means;
    Digital processing means for generating a noise reduced audio signal using the digital audio signal and the selected set of parameters;
    A speaker for sound reproduction and the noise reducing audio signal, and said predetermined input sound signal,
    With headphones.
  2. At the time of turning on the power, when there is a predetermined operation input, at regular time intervals, or in all cases,
    The noise characteristic analysis means analyzes the characteristic of the noise,
    Said selection and setting means, on the basis of the analysis result of the noise characteristic analyzing means, to select a set of one parameter from among the multiple sets of parameters stored in said holding means, said supplied to the digital processing unit Update parameters,
    The headphones according to claim 1.
  3. Means for extracting noise by removing a component of the predetermined input audio signal from the audio signal picked up by the microphone;
    The headphone according to claim 1, wherein the noise characteristic analysis unit analyzes a characteristic of the noise extracted by the noise extraction unit.
  4. When a change in noise obtained by collecting with the microphone becomes a predetermined magnitude or more, the noise characteristic analysis means analyzes the characteristic of the noise,
    Said selection and setting means, on the basis of the analysis result of the noise characteristic analyzing means, to select a set of one parameter from among the multiple sets of parameters stored in said holding means, said supplied to the digital processing unit The headphones according to any one of claims 1 to 3, wherein parameters are updated.
  5. Providing analog processing means for generating a second noise-reduced audio signal from the noise analog audio signal;
    The system further comprising: a system that acoustically reproduces the second noise-reduced audio signal generated by the analog processing unit and acoustically synthesizes the second noise-reduced audio signal with the noise to reduce the noise. Item 5. The headphone according to any one of Items 4.
  6. 6. The speaker according to claim 1, wherein when the speaker is worn, an audio signal obtained by adding the noise-reduced audio signal and the predetermined input audio signal by an adding unit is added near the ear. Headphones as described in any one.
  7. An analog audio signal collected by a microphone installed outside the headphone housing and collecting external noise is converted into a digital audio signal by A / D conversion means,
    When muting a predetermined input audio signal,
    The digital audio signal obtained by the A / D conversion means is analyzed for noise characteristics after high-frequency component removal and downsampling .
    Held by the holding means, among the plurality of kinds of a plurality of sets prepared in advance in accordance with the noise characteristic parameter, select a set of parameters suitable for the reduction of noise is a noise characteristic of the result of said analysis,
    Using the digital audio signal and the set of selected parameters to generate a noise reduced audio signal;
    Acoustic reproduction and the noise reducing audio signal, and said predetermined input sound signal from the speaker,
    Noise reduction method for headphones.
  8. An analog audio signal obtained by collecting sound from a microphone installed outside the headphone housing and collecting external noise is converted into a digital audio signal, and the noise is reduced from the digital audio signal. As a calculation process for generating a noise-reduced audio signal, acoustically reproducing the noise-reduced audio signal from a speaker, and acoustically synthesizing with the noise to reduce the noise,
    When muting a predetermined input audio signal,
    Noise characteristic analysis processing for analyzing noise characteristics after performing high-frequency component removal and downsampling for digital audio signals of noise obtained by collecting with the microphone,
    Parameters used for generating the noise-reduced speech signal, and a holding unit that holds a plurality of sets of parameters prepared in advance according to a plurality of types of noise characteristics. A selection setting process for selecting a set of parameters suitable for reducing noise, which is a noise characteristic;
    Receiving the digital audio signal and receiving the set of parameters selected in the selection setting process to generate the noise-reduced audio signal; and
    An audio reproduction process for synthesizing the noise-reduced audio signal and a predetermined input audio signal for audio reproduction by the speaker;
    Is a noise reduction processing program that causes an arithmetic processing unit to execute.
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JPH05188977A (en) * 1992-01-17 1993-07-30 Matsushita Electric Ind Co Ltd Noise controller
JPH0659689A (en) * 1992-08-10 1994-03-04 Fujitsu Ten Ltd Noise control device
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JPH0756579A (en) * 1993-06-30 1995-03-03 Ricoh Co Ltd Noise control device
JPH07104767A (en) * 1993-10-04 1995-04-21 Toyota Motor Corp Device for reducing noise in car
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