JP5488128B2 - Signal processing device - Google Patents

Description

  The present invention relates to a technique for calculating frequency characteristics of an audio device.

  In acoustic devices such as a microphone and a speaker, each frequency characteristic varies depending on individual differences, and the characteristic may change with time. Therefore, there are cases where it is desired to examine the frequency characteristics of the audio equipment when adjusting the sound field characteristics. For example, Patent Document 1 describes that the frequency characteristics of a speaker are corrected using a speaker and a microphone in order to obtain a desired acoustic characteristic of space.

JP 2008-227680 A

By the way, since both the microphone and the speaker have unique frequency characteristics, it is difficult for the technique described in Patent Document 1 to independently obtain the frequency characteristics for each device without being affected by each other.
The present invention has been made in view of the above-described problems, and an object of the present invention is to provide a technique that makes it possible to calculate each frequency characteristic while suppressing the mutual influence of a microphone and a speaker.

  In order to achieve the above-described object, a signal processing apparatus according to the present invention provides a sound source that supplies a sound signal, a first connection unit to which a microphone that picks up a sound by vibrating a vibration surface, and a supply from the sound source. A second connecting portion to which a speaker that emits a sound with a vibration surface larger than the vibration surface vibrated according to a sound signal to be emitted, and a sound emitted by the speaker according to a predetermined sound signal supplied by the sound source When the impulse sound is picked up by the microphone connected to the first connecting portion and arranged at a predetermined distance from the speaker, the response specifying means for specifying the impulse response based on the sound pickup result And the time-series change of the impulse response specified by the response specifying means is divided into one corresponding to the microphone and one corresponding to the speaker according to the distance. Characterized in that it comprises a separating means for.

  In a preferred aspect, the first calculating means for calculating the frequency characteristic of the microphone based on the impulse response separated by the separating means as corresponding to the microphone, and the separating means corresponding to the speaker by the separating means. A second calculating means for calculating a frequency characteristic of the speaker based on the impulse response, a first frequency characteristic that is a reference frequency characteristic of the microphone, and a second frequency characteristic that is a reference characteristic of the speaker. Storage means for storing frequency characteristics, first adjustment means for adjusting a sound signal representing sound collected by the microphone, second adjustment means for adjusting a sound signal supplied from the sound source, and The first adjusting means makes the frequency characteristic calculated by one calculating means approach the first frequency characteristic. Correcting the adjustments, the calculated frequency characteristic by said second calculating means so as cause closer to the second frequency characteristic, and a correcting means for correcting the adjustment contents of the second adjusting means.

  In another preferred embodiment, the second calculation means is based on a frequency characteristic calculated from the impulse response specified by the response specifying means and an inverse characteristic of the frequency characteristic of the microphone calculated by the first calculation means. Then, the frequency characteristic of the speaker is calculated.

  According to the present invention, it is possible to calculate each frequency characteristic while suppressing the mutual influence of the microphone and the speaker.

Block diagram showing the configuration of the sound processor The block diagram which shows the structure of the signal processing part of 1st Embodiment. Graph explaining impulse sound and its impulse response Diagram explaining impulse response The block diagram which shows the structure of the signal processing part of 2nd Embodiment. A graph explaining the processing of the correction unit

Hereinafter, embodiments of the present invention will be described with reference to the drawings.
[First Embodiment]
First, a first embodiment of the present invention will be described.
FIG. 1 is a block diagram showing a hardware configuration of a sound processing apparatus 1 according to this embodiment.
Here, the sound processing device 1 is a karaoke device. As shown in FIG. 1, the sound processing device 1 includes a control unit 11, an operation unit 12, a display unit 13, a storage unit 14, and a signal processing unit 15. The microphone 16 and the speaker 17 shown in FIG. 1 can be attached to and detached from the signal processing unit 15, respectively.
The control unit 11 includes a central processing unit (CPU), a read only memory (ROM), and a random access memory (RAM). The CPU executes programs stored in the ROM and the storage unit 14 using the RAM as a work area, and controls each unit of the sound processing device 1. The operation unit 12 is, for example, a remote controller or an operator of the apparatus main body. When an operation is performed by a user, the operation unit 12 outputs an operation signal corresponding to the operation content to the control unit 11. The display unit 13 is a display unit including, for example, a liquid crystal display, and displays various types of information including image images and lyrics telop relating to karaoke accompaniment. The storage unit 14 corresponds to an example of a storage unit including a hard disk device, for example, and stores various programs related to karaoke accompaniment such as a program for operating the control unit 11 and music data and lyrics telop. In addition to this, the storage unit 14 stores a “first reference characteristic” that is a frequency characteristic that is a reference of the microphone 16 and a “second reference characteristic” that is a frequency characteristic that is a reference of the speaker 17. The signal processing unit 15 corresponds to the signal processing device of the present invention. The signal processing unit 15 is, for example, a DSP (Digital Signal Processor), and performs various types of signal processing in accordance with the control of the control unit 11.
In addition to the above, the sound processing device 1 has a function that a karaoke device normally has. For example, the sound processing device 1 has a communication means, and acquires music data from an external device connected via a communication line. In addition, the frequency characteristic of this embodiment means a frequency characteristic of a sound pressure level unless otherwise specified.

FIG. 2 is a block diagram illustrating a configuration of the signal processing unit 15.
The first connection portion 151A is a connection terminal to which the microphone 16 is connected, and is, for example, a jack into which the plug of the microphone 16 is inserted. The second connection portion 151B is a connection terminal to which the speaker 17 is connected, for example, a jack into which the speaker 17 is plugged.
The microphone 16 that can be connected to the signal processing unit 15 has a diaphragm that is a vibrating body like a moving coil type, and the diaphragm receives a sound wave and vibrates to obtain a sound signal that is an electric signal. The microphone 16 outputs an analog sound signal representing the collected sound to the signal processing unit 15. The speaker 17 that can be connected to the signal processing unit 15 has a vibrating body (diaphragm) like a dynamic type. When a sound signal, which is an electrical signal, is supplied to the speaker 17, the diaphragm vibrates and emits sound according to the sound signal.

  The vibration surface of the diaphragm included in the speaker 17 is larger than the vibration surface of the diaphragm included in the microphone 16. The diaphragm vibration surface of the microphone 16 is a surface of the diaphragm facing the space in which sound waves propagate. The vibration surface of the diaphragm included in the speaker 17 is a surface facing a space in which sound waves are emitted by the vibration of the diaphragm.

The A (Analog) / D (Digital) converter 152A, when a sound signal is supplied from the microphone 16 via the first connection unit 151A, converts the sound signal into a digital sound signal and outputs the sound signal. is there.
The first equalizer 153A corresponds to an example of a first adjustment unit that electrically adjusts the sound signal supplied from the A / D converter 152A. The first equalizer 153A is controlled by the control unit 11 in accordance with the operation content of the operation unit 12 of the user, and performs adjustment to give a sound effect to the sound signal. The first equalizer 153A is for adjusting the sound signal output from the microphone 16, and has a function of adjusting the sound pressure level of each frequency band represented by at least the sound signal.

The sound source 154 supplies a sound signal representing various sounds such as karaoke music and a measurement sound signal. The measurement sound signal is a predetermined sound signal for causing the speaker 17 to emit an impulse sound.
The mixer 155 synthesizes and outputs the sound signal adjusted by the first equalizer 153A and the sound signal supplied from the sound source 154.

  The second equalizer 153B corresponds to an example of a second adjustment unit that electrically adjusts the sound signal supplied from the mixer 155. The second equalizer 153 </ b> B is adjusted by the control unit 11 in accordance with the operation content of the operation unit 12 of the user, and thereby adjusts the sound effect to the sound signal. The second equalizer 153B adjusts the sound signal supplied from the sound source 154 and has a function of adjusting the sound pressure level of each frequency band represented by at least the sound signal. In other words, the second equalizer 153B uses the sound signal supplied from the sound source 154 to the speaker 17 without passing through the microphone 16 as an adjustment target.

When a sound signal is supplied from the second equalizer 153B, the D (Digital) / A (Analog) converter 152B converts the sound signal into an analog sound signal and outputs it. The sound signal output from the D / A converter 152B is supplied to the speaker 17 via the second connection unit 151B.
Note that a preamplifier and a power amplifier are actually provided between the D / A converter 152B and the speaker 17, but the description thereof is omitted here.
Functions related to karaoke accompaniment are realized by the above-described configuration of the signal processing unit 15. Next, a configuration for realizing functions other than those related to karaoke accompaniment will be described.

The response specifying unit 156 corresponds to an example of a response specifying unit of the present invention, and operates when an impulse sound is emitted from the speaker 17 in accordance with a measurement sound signal supplied from the sound source 154. When the impulse sound is collected by the microphone 16, the response identifying unit 156 identifies the impulse response based on the sound collection result.
At this time, it is assumed that both are arranged so that the distance between the microphone 16 and the speaker 17 is a predetermined distance. With regard to this arrangement, the sound collection surface of the microphone 16 may be directed to the sound emission surface of the speaker 17 and the positional relationship between the two may be determined using a jig, for example.

Here, the impulse sound emitted from the speaker 17 and the impulse response will be specifically described. FIG. 3 is a graph showing a time-series change in the amplitude of the impulse sound and the impulse response thereof. FIG. 3A shows a time-series change in the amplitude of the impulse sound. FIG. 3B shows a time-series change in the amplitude of the impulse response. 3 (a) and 3 (b), the horizontal axis represents the time axis.
As shown in FIG. 3A, the impulse sound is represented by a sound wave whose amplitude suddenly increases in a short period of time. Hereinafter, the period during which the impulse sound is emitted is referred to as “impulse period Δti”. Further, as shown in FIG. 3A, even after the impulse period Δti, a sound wave having a non-zero amplitude continues to be emitted. This occurs because the diaphragm continues to vibrate even after the impulse sound is emitted during the impulse period Δti. That is, even after the diaphragm is vibrated by emitting desired impulse sound, the diaphragm continues to vibrate for a while due to the reverberation. In this way, a period in which the reverberation sound of the impulse sound is emitted after the impulse period Δti is hereinafter referred to as a “reverberation period Δtr”.

Next, as shown in FIG. 3B, when the time at which the impulse sound during the impulse period Δti is emitted is “0”, the amplitude of the impulse response increases after time td. More specifically, the impulse response to the impulse sound during the impulse period Δti appears in the first time range Δta1 starting from time td. In the first time range Δta1, many impulse responses having a relatively large amplitude appear in a short period of time.
The time td is determined by the distance between the speaker 17 and the microphone 16, and may be considered to be determined by the time obtained by dividing the distance by the speed of sound.

As described above, an impulse response to the impulse sound in the impulse period Δti appears in the first time range Δta1, but an impulse response also appears in the second time range Δta2 that follows. The impulse response in the second time range Δta2 appears due to the impulse sound in the reverberation period Δtr. That is, since the reverberant sound continues to be emitted even after the desired impulse sound, the response appears in the second time range Δta2.
The response specifying unit 156 specifies an impulse response in the time range Δta including at least the first time range Δta1 and the second time range Δta2.

The separation unit 160 corresponds to an example of the separation unit of the present invention, and corresponds to the microphone 16 with respect to the time series change of the impulse response specified by the response specifying unit 156 according to the distance between the microphone 16 and the speaker 17. And those corresponding to the speaker 17 are separated on the time axis.
The first calculator 157 corresponds to an example of a first calculator of the present invention, and calculates the frequency characteristic of the microphone 16 based on the impulse response in the first time range Δta1 separated by the separator 160.
The second calculator 158 corresponds to an example of the second calculator of the present invention, and calculates the frequency characteristic of the speaker 17 based on the impulse response in the second time range Δta2 separated by the separator 160. Specifically, the second calculation unit 158 is based on the frequency characteristic calculated from the impulse response in the time range Δta and the inverse characteristic of the frequency characteristic of the microphone 16 obtained from the impulse response in the first time range Δta1. The frequency characteristic of the speaker 17 is calculated. This inverse characteristic is a characteristic to be multiplied by the frequency characteristic of the microphone 16 in order to make the frequency characteristic of the microphone 16 flat (make the sound pressure level constant at each frequency). The second calculation unit 158 calculates the frequency characteristic of the speaker 17 after canceling the frequency characteristic of the microphone 16 by calculating the frequency characteristic calculated from the impulse response in the time range Δta and the inverse characteristic of the frequency characteristic of the microphone 16. To do.

FIG. 4 is a diagram for explaining an impulse response used by the first calculation unit 157 and the second calculation unit 158 for calculating frequency characteristics. In the graph shown in FIG. 4, the horizontal axis is the time axis, and the impulse response shown in FIG.
As shown in FIG. 4, the first calculation unit 157 performs a well-known fast Fourier transform (FFT) using a short-term impulse response of the first time range Δta1 after the impulse response appears, Perform frequency analysis. The first time range Δta1 is, for example, 1 millisecond from the time td. The second calculation unit 158 refers to the impulse response in the time range Δta including the second time range Δta2, and performs frequency analysis by performing a known fast Fourier transform using the impulse response in the time range Δta. The time range Δta is, for example, 10 milliseconds from the time td.

Next, the reason why the impulse response of the first time range Δta1 is used when calculating the frequency characteristic of the microphone 16 and the impulse response of the second time range Δta2 is used when calculating the frequency characteristic of the speaker 17 will be described.
As described above, the vibration surface of the diaphragm of the speaker 17 is larger than the vibration surface of the diaphragm of the microphone 16, and when the impulse sound is emitted, the vibration plate of the speaker 17 extends over a relatively wide time range. Vibrate. The diaphragm of the microphone 16 also vibrates due to the collection of the impulse sound. In this case as well, the diaphragm 16 is divided into a period corresponding to the impulse sound during the impulse period Δti and a period corresponding to the reverberant sound during the reverberation period Δtr. It is done. Among these, the vibration period corresponding to the impulse period Δti roughly corresponds to the first time range Δta1, and the vibration period corresponding to the reverberation period Δtr roughly corresponds to the second time range Δta2.

  As described above, since the vibration surface of the speaker 17 is larger than the vibration surface of the microphone 16, the vibration period of the speaker 17 is longer than that of the microphone 16, and the impulse response depending on the speaker 17 is in the second time range. Disperses over a wide range like Δta2. On the other hand, since the diaphragm of the microphone 16 is small, the vibration period is short, and the impulse response depending on the microphone 16 appears in a concentrated manner in the first time range Δta1. Therefore, in the second time range Δta2, the ratio of the impulse response generated by the vibration of the vibration surface of the speaker 17 is considered to be larger than the ratio of the impulse response due to the vibration of the vibration surface of the microphone 16.

For the above reason, the separation unit 160 separates the impulse response in the first time range Δta1 as corresponding to the microphone 16, and the first calculation unit 157 calculates the frequency characteristic based on the separated impulse response. The influence of the second time range Δta <b> 2 having a large impulse response depending on the speaker 17 can be excluded. Therefore, the first calculation unit 157 can calculate the frequency characteristic of the microphone 16 more accurately while suppressing the influence of the speaker 17.
The separation unit 160 separates the impulse response of the time range Δta including the second time range Δta2 as corresponding to the speaker 17, and the second calculation unit 158 calculates the frequency characteristics based on the separated impulse response. To do. In this case, the second calculation unit 158 refers to the second time range Δta2 in which many impulse responses depending on the speaker 17 are included. Therefore, the second calculation unit 158 can calculate the frequency characteristic of the speaker 17 while suppressing the influence of the microphone 16.

The first calculator 157 and the second calculator 158 output information indicating the calculated frequency characteristics to the controller 11.
The control unit 11 causes the display unit 13 to display a graph that compares the first reference characteristic stored in the storage unit 14 with the calculated frequency characteristic of the microphone 16. Further, the control unit 11 causes the display unit 13 to display a graph that compares the second reference characteristic stored in the storage unit 14 with the calculated frequency characteristic of the speaker 17. The user who sees the display content of the display unit 13 determines how much the microphone 16 and the speaker 17 differ from the reference characteristics. And if it judges that both characteristics differ greatly, a user will operate the operation part 12, and will adjust the acoustic effect which 1st equalizer 153A and 2nd equalizer 153B provide with respect to a sound signal, and the desired characteristic To be. If the difference between the two characteristics is so large that adjustment is difficult, the user takes necessary measures such as maintaining the microphone 16 and the speaker 17 or replacing them with new ones.
In addition, the control unit 11 accumulates the difference in sound pressure level between the calculated frequency characteristic and the first and second reference characteristics in a predetermined frequency width unit, and when the accumulated value is equal to or greater than a threshold value. , “Please replace the microphone.” “Speaker is degraded” messages may be displayed on the display unit 13 to notify information about the frequency characteristics of the microphone 16 and the speaker 17.
The contents of this notification are merely examples, and the contents are not limited to those described above.

As in the first embodiment described above, the time range in which the impulse response appears varies depending on the magnitude relationship between the vibration surfaces of the microphone 16 and the speaker 17. Using this, the first calculation unit 157 calculates the frequency characteristic of the microphone 16 based on the impulse response in the first time range Δta1, and the second calculation unit 158 calculates the speaker 17 based on the impulse response in the second time range Δta2. Calculate the frequency characteristics. With this configuration, it is possible to independently obtain the frequency characteristics of the microphone and the speaker while suppressing the mutual influence of the audio equipment.
The second calculation unit 158 calculates the frequency characteristic of the speaker 17 based on the frequency characteristic of the microphone 16 calculated from the first time range Δta1 and the inverse characteristic of the frequency characteristic calculated from the time range Δta. To do. Thereby, in the sound processing apparatus 1, when calculating the frequency characteristics of the microphone 16 and the speaker 17, it is only necessary to emit the impulse sound only once. Therefore, compared with the case where a plurality of impulse sounds are required, the time required for the processing relating to the calculation of the frequency characteristic can be shortened.

[Second Embodiment]
Next, a second embodiment of the present invention will be described.
The overall hardware configuration of the sound processing apparatus 1 of this embodiment is the same as that of the first embodiment described above. In this embodiment, the signal processing unit has a function of spontaneously correcting the adjustment contents of the first equalizer 153A and the second equalizer 153B based on the calculated frequency characteristics of the microphone 16 and the speaker 17.
In the following description, among the configurations included in the karaoke apparatus according to this embodiment, the same configurations as those included in the sound processing device 1 according to the first embodiment are denoted by the same reference numerals and description of those configurations is omitted. To do.

FIG. 5 is a block diagram showing a hardware configuration of the signal processing unit 15a of this embodiment.
As shown in FIG. 5, the signal processing unit 15 a has a correction unit 159 in addition to the same configuration as that of the first embodiment described above.
The correcting unit 159 is an example of a correcting unit according to the present invention, and corrects the adjustment content of the first equalizer 153A based on the frequency characteristic of the microphone 16 calculated by the first calculating unit 157. Further, the correction unit 159 corrects the adjustment content of the second equalizer 153B based on the frequency characteristic of the speaker 17 calculated by the second calculation unit 158.

FIG. 6 is a diagram for explaining the correction process performed by the correction unit 159.
For example, the frequency characteristic and the first reference characteristic of the microphone 16 are as shown in FIG. In FIG. 6A, the horizontal axis represents frequency, and the vertical axis represents sound pressure level. The dotted line represents the first reference characteristic of the microphone 16, and the solid line represents the frequency characteristic of the microphone 16 calculated by the first calculation unit 157.
The correction unit 159 specifies a difference in sound pressure level between the frequency characteristic calculated by the first calculation unit 157 and the reference characteristic of the microphone 16 for each predetermined frequency width, and reduces the difference by the first equalizer. The adjustment content of 153A is corrected. Specifically, the correction unit 159 lowers the sound pressure level in the frequency range fm1 so as to bring the frequency characteristic indicated by the solid line in FIG. 6A closer to the first reference characteristic indicated by the dotted line, so that the frequency characteristic indicated by the solid line in FIG. The first equalizer 153A is adjusted to increase the sound pressure level.

Further, it is assumed that the frequency characteristic and the second reference characteristic of the speaker 17 are as shown in FIG. In FIG. 6B, the horizontal axis represents frequency, and the vertical axis represents sound pressure level. The dotted line represents the second reference characteristic of the speaker 17, and the solid line represents the frequency characteristic of the speaker 17 calculated by the second calculation unit 158.
The correction unit 159 specifies a difference in sound pressure level between the frequency characteristic calculated by the second calculation unit 158 and the reference characteristic of the microphone 16 for each predetermined frequency width, and reduces the difference by using the second equalizer. The adjustment content of 153B is corrected. Specifically, the correction unit 159 increases the sound pressure level in the frequency range fs1 so as to bring the frequency characteristic indicated by the solid line in FIG. 6B closer to the second reference characteristic indicated by the dotted line, and increases the sound pressure level in the frequency range fs2. The second equalizer 153B is adjusted so as to lower the sound pressure level.

After the correction by the correction unit 159, the first equalizer 153A generates a sound signal representing the sound collected by the microphone 16 so that the frequency characteristic calculated by the first calculation unit 157 approaches the first reference characteristic. It will be adjusted. In addition, the second equalizer 153B adjusts the sound signal supplied from the sound source 154 so that the frequency characteristic calculated by the second calculation unit 158 approaches the second reference characteristic.
According to the second embodiment described above, since the microphone 16 and the speaker 17 are adjusted to suitable frequency characteristics without being conscious of the user, the burden on the adjustment of the frequency characteristic imposed on the user can be reduced.

[Modification]
The present invention can be implemented in a form different from the above-described embodiment. Further, the following modifications may be combined as appropriate.
(Modification 1)
In each embodiment mentioned above, the signal processing apparatus of this invention was applied to the signal processing part of the sound processing apparatus 1 which is a karaoke apparatus. In addition to this, the signal processing apparatus of the present invention can be applied to a general-purpose computer apparatus, or may be applied to, for example, an inspection apparatus that inspects frequency characteristics of a microphone and a speaker. In this inspection apparatus, when a microphone is connected to the first connection terminal and a speaker is connected to the second connection terminal, and a predetermined operation is performed by the user, an impulse sound is output from the speaker according to the measurement sound signal from the sound source. Sounds out. At this time, it is preferable that the inspection apparatus is configured so that the distance between the microphone and the speaker is a predetermined distance. Then, when the signal processing unit calculates the frequency characteristics of the microphone and the speaker according to the algorithm of the above-described embodiment, information indicating the inspection result is notified. By this notification, the user can know the state of the device such as whether or not the microphone and the speaker are deteriorated.
Depending on the apparatus such as the above-described inspection apparatus, the one corresponding to the first equalizer 153A and the second equalizer 153B is not necessary.

(Modification 2)
In the second embodiment described above, the sound processing device 1 emits the impulse sound only once and calculates the frequency characteristics of the microphone 16 and the speaker 17, respectively. The configuration of the second embodiment described above may be modified as follows.
First, a measurement sound signal is supplied from the sound source 154, and a first impulse sound is emitted from the speaker 17. When the first calculation unit 157 calculates the frequency characteristic of the microphone 16, the correction unit 159 corrects the adjustment content of the first equalizer 153A based on the inverse characteristic of the frequency characteristic so that the characteristic becomes flat. To. Thereby, first, the frequency characteristic of the microphone 16 is adjusted to be the first reference characteristic. Subsequently, a measurement sound signal is supplied from the sound source 154, and a second impulse sound is emitted from the speaker 17. Then, the second calculation unit 158 obtains the frequency characteristic of the speaker 17 based on the impulse response in the time range Δta specified by the response specifying unit 156. Then, the correction unit 159 corrects the adjustment content of the second equalizer 153B based on the second reference characteristic. In this case, since the frequency characteristic of the microphone 16 has already been adjusted to be the first reference characteristic, the influence of the frequency characteristic of the microphone 16 can be ignored. Even with this configuration, the signal processing unit 15a can calculate the frequency characteristics while suppressing the mutual influence of the devices, as in the second embodiment.
As described above, the sound processing device 1 may emit an impulse sound a plurality of times for calculating the frequency characteristics of the microphone 16 and the speaker 17. In addition, the configuration of this modification may be combined with the configuration of the first embodiment, and the user himself / herself may make an adjustment corresponding to the correction process executed by the correction unit 159.

(Modification 3)
In each of the above-described embodiments, the first and second reference characteristics represent flat characteristics having the same sound pressure level at each frequency, but what are the characteristics of the first and second reference characteristics? There may be. For example, if importance is attached to the frequency characteristics of the low sound, the sound pressure level may be increased as the frequency is lower, and the sound pressure level may be decreased as the frequency is higher.

(Modification 4)
Further, the signal processing device of the present invention may be applied to a device for controlling sound field characteristics (hereinafter referred to as “sound field control device”).
For example, the sound field control device emits an impulse sound to each of a plurality of speakers according to the measurement sound signal when adjusting the sound field characteristics before the actual performance. When the impulse sound emitted from the speaker is picked up by the microphone, the sound field control device converts the frequency characteristics of the sound pressure level of the speaker and the microphone based on the sound signal representing the picked up sound. As specified.

Next, the correction unit 159 compares the specified sound field characteristic with a sound field characteristic that is a reference in the sound field (hereinafter referred to as “reference sound field characteristic”), and converts the sound field characteristic into the reference sound field. In order to approximate the characteristics, the frequency characteristics of each microphone and speaker are adjusted. Again, the correction unit 159 corrects the adjustment content of the adjustment means that is the first equalizer to adjust the frequency characteristic of the microphone, and the adjustment content of the adjustment means that is the second equalizer to adjust the frequency characteristic of the speaker. Correct. The reference sound field characteristic is determined in advance, and data representing it is stored in the storage unit 14. The reference sound field characteristics need only be determined to represent preferable sound field characteristics, and are based on sound field conditions such as sound field type, music type, performance content, and PA engineer's preference. is there. When the sound field condition including the type of the sound field is input by the operation of the operation unit 12 of the PA engineer, the correction unit 159 approaches the reference sound field characteristic according to the input sound field condition. Control the sound field characteristics. Here, the correction unit 159 compares the frequency characteristic of the sound pressure level of the specified sound field characteristic with the frequency characteristic of the sound pressure level of the reference sound field characteristic determined for each frequency component, and there is a difference between the two. The adjustment content of each adjustment means is corrected so as to be reduced, and the frequency characteristics of the microphone and the speaker are adjusted.
Note that the sound field characteristic to be controlled by the correction unit 159 may be other than the frequency characteristic of the sound pressure level, for example, a reverberation characteristic. In this case, the correction unit 159 may be controlled to reduce the sound pressure level in the low frequency band so as to shorten the reverberation time that is too long in the low frequency band, for example. Further, when it is desired to locally reduce the sound of a certain frequency at a certain location, the correction unit 159 lowers the sound pressure level of the frequency with respect to the frequency characteristics of the speaker near the certain location. The adjustment content of the adjustment means may be corrected as described above.

  According to the above sound field control device, the individual acoustic characteristics for each sound field can be controlled to be predetermined characteristics by automatic control of the adjusting means. In addition, the control of the correction unit 159 adjusts the reference sound field characteristics regardless of the conditions for each sound field. Therefore, for a PA engineer, the difference in adjustment contents is reduced depending on the sound field characteristics. Therefore, the PA engineer can easily adjust the mixer to adjust the sound field characteristics according to his / her preference compared to the case where there is no automatic control by the sound field control device. It becomes easier for PA engineers to adjust the sound field characteristics.

(Modification 5)
In the first embodiment described above, the signal processing unit 15 includes the first calculation unit 157 and the second calculation unit 158 in order to calculate the frequency characteristics based on the impulse response separated by the separation unit 160. It was. Instead of this, the configuration of the first calculation unit 157 and the second calculation unit 158 of the signal processing unit 15 is omitted, and for example, the separation unit 160 outputs information representing the separation result of the impulse response to the outside of the signal processing unit 15. May be. In this case, it is preferable that an external device (for example, the control unit 11) as viewed from the signal processing unit 15 realizes functions corresponding to the first calculation unit 157 and the second calculation unit 158.
The first time range Δta1 and the second time range Δta2 described above describe an algorithm for separating the time-series change of the impulse response into those corresponding to the microphone 16 and those corresponding to the speaker 17, respectively. It was used for this purpose. Therefore, the separation unit 160 is not limited to the configuration that separates the impulse responses according to the time range divided in advance. For example, the separation unit 160 may separate the impulse responses with reference to the amplitude of the impulse response or the period of appearance. Good.

(Modification 6)
The programs executed by the control unit 11 and the signal processing units 15 and 15a of the sound processing device 1 in each of the above-described embodiments are a magnetic recording medium (magnetic tape, magnetic disk, etc.), an optical recording medium (optical disk, etc.), and a magneto-optical device. It can be provided in a state where it is recorded on a computer-readable recording medium such as a recording medium or a semiconductor memory. It is also possible to download the sound processing apparatus 1 via a network such as the Internet.

DESCRIPTION OF SYMBOLS 1 ... Sound processing apparatus, 11 ... Control part, 12 ... Operation part, 13 ... Display part, 14 ... Memory | storage part, 15 and 15a ... Signal processing part, 151A ... 1st connection part, 151B ... 2nd connection part, 153A ... 1st equalizer, 153B ... 2nd equalizer, 154 ... Sound source, 155 ... Adder, 156 ... Response specifying part, 157 ... 1st calculation part, 158 ... 2nd calculation part, 159 ... Correction part, 160 ... Separation part, 16 ... microphone, 17 ... speaker.

Claims (3)

A sound source for supplying sound signals;
A first connection portion to which a microphone that vibrates and collects sound is connected;
A second connection unit to which a speaker that vibrates and emits a vibration surface larger than the vibration surface according to a sound signal supplied from the sound source;
An impulse sound emitted by the speaker in response to a predetermined sound signal supplied by the sound source is collected by the microphone connected to the first connection unit and disposed at a predetermined distance from the speaker. And a response specifying means for specifying an impulse response based on the sound collection result,
Separating means for separating the time-series change of the impulse response specified by the response specifying means into one corresponding to the microphone and one corresponding to the speaker according to the distance. Signal processing device. First calculating means for calculating a frequency characteristic of the microphone based on the impulse response separated as corresponding to the microphone by the separating means;
Second calculating means for calculating a frequency characteristic of the speaker based on the impulse response separated as corresponding to the speaker by the separating means;
Storage means for storing a first frequency characteristic which is a frequency characteristic which is a reference of the microphone and a second frequency characteristic which is a frequency characteristic which is a reference of the speaker;
First adjusting means for adjusting a sound signal representing sound collected by the microphone;
Second adjusting means for adjusting a sound signal supplied from the sound source;
The adjustment content of the first adjustment unit is corrected so that the frequency characteristic calculated by the first calculation unit is close to the first frequency characteristic, and the frequency characteristic calculated by the second calculation unit is changed to the second frequency. The signal processing apparatus according to claim 1, further comprising: a correction unit that corrects the adjustment content of the second adjustment unit so as to approximate the characteristic. The second calculation means includes
The frequency characteristic of the speaker is calculated based on the frequency characteristic calculated from the impulse response specified by the response specifying means and the inverse characteristic of the frequency characteristic of the microphone calculated by the first calculating means. The signal processing apparatus according to claim 2.

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