JP2007114239A - Device and method for controlling musical sound - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To improve singing ability better than actual by improving tone quality. <P>SOLUTION: A musical sound controller includes: a microphone element 5 for outputting a voice signal according to a voice input to a microphone 1; a distance sensor 6 for detecting the distance between the microphone 1 and the mouth of a singer whose voice is to be input; and a CPU and DSP part which apply effect processing to the voice signal outputted from the microphone element 5 according to the distance detected by the distance sensor 6. <P>COPYRIGHT: (C)2007,JPO&INPIT

Description

  The present invention relates to a musical tone control device and a musical tone control method, and more particularly to a musical tone control device and a musical tone control method for controlling an audio signal output from a microphone.

In a widely used karaoke system, a person who sings inputs voice to a microphone in accordance with an accompaniment. Accompaniment music is generated by an automatic performance device. In recent karaoke systems, there are products in which this automatic performance device has various functions. There have also been some proposals regarding the functions of automatic performance devices. For example, in a proposed automatic performance device with a singing ability scoring function, when singing loudly at karaoke, a person usually does not raise his voice, and the microphone can be brought closer to the mouth without changing the volume of the voice. Many. Therefore, if the distance between the microphone and the singer's mouth is determined, the singer can indirectly determine whether or not the singer is singing at a certain volume. When singing, it detects the distance between the microphone and the singer's mouth and the volume value of the melody data that serves as a pre-stored example of singing for each melody data. The singing ability of the singer is scored by comparing the magnitude of the volume of the sound and the distance between the microphone and the singer's mouth (see Patent Document 1).
JP-A-5-181407

However, in the conventional techniques including the above-mentioned Patent Document 1, it is possible to score the singing ability, but the singing ability is not improved more than the actual one. Karaoke that is widespread like today is used for events such as relatives, banquets and entertainment in organizations such as companies rather than competing for singing ability. However, it is embarrassing to sing in front of everyone for those who do not have singing ability, and relatives, banquets, entertainment etc. accompanied by karaoke events are painful. On the other hand, in the case of a person who has a strong desire for self-revelation, it is often the case that a professional singer is imitated with a mouth too close to a microphone or sings loudly. When the mouth is brought too close to the microphone, unpleasant sounds such as breath noise (breathing noise), sound due to the viscosity of saliva in the oral cavity, and sound that the tongue is swallowed in the oral cavity are generated. Also, singing out loud can cause discomfort to those who are listening.
An object of the present invention is to solve such a conventional problem, and it is an object of the present invention to improve the singing ability by improving the sound quality by controlling an audio signal output from a microphone. Furthermore, it aims at suppressing breath noise and an unpleasant sound in the oral cavity by controlling the audio signal output from the microphone.

  The musical tone control apparatus according to claim 1 includes a signal generating means (in the embodiment, corresponding to the microphone element 5 in FIG. 2) for outputting a sound signal in accordance with the sound input to the microphone, the microphone and the sound input. Distance detection means (in the embodiment, corresponding to the distance sensor 6 in FIG. 2 and the CPU 10 in FIG. 4) and the distance detected by the distance detection means. And a signal processing means (in the embodiment, corresponding to the CPU 10 and the DSP unit 18 in FIG. 4) for effecting the audio signal output from the signal generating means.

  In the musical tone control apparatus according to claim 1, as described in claim 2, the signal processing means may be configured to be accommodated in the microphone.

  3. The musical tone control apparatus according to claim 1, wherein the distance detecting means irradiates light toward the outside of the microphone and detects the distance based on the intensity of the reflected light, and the signal processing means. Generates a coefficient of effect processing (corresponding to a coefficient input from the CPU 10 in FIG. 4 to the DSP unit 18 in FIG. 4) according to the distance detected by the distance detecting means, and a signal is generated by the generated coefficient. The audio signal output from the generating unit may be controlled.

4. The musical tone control apparatus according to claim 3, wherein the signal processing means controls the frequency / gain characteristics of the audio signal output from the signal generating means by the generated coefficient. Good.
Alternatively, in the musical tone control apparatus according to claim 3, as described in claim 5, the signal processing means is configured to control a cutoff frequency of the audio signal output from the signal generating means by the generated coefficient. Also good.
Alternatively, in the musical tone control apparatus according to claim 3, as described in claim 6, the signal processing means controls the Q factor which is the steepness of resonance of the audio signal output from the signal generating means by the generated coefficient. Such a configuration may be adopted.
Alternatively, in the musical tone control apparatus according to claim 3, as described in claim 7, the signal processing means controls the output level of the audio signal output from the signal generating means by the generated coefficient. Good.
Alternatively, in the musical tone control device according to claim 3, as described in claim 8, when the level of the audio signal output from the signal generation unit is equal to or less than a predetermined threshold, the signal processing unit A configuration may be adopted in which a coefficient for greatly attenuating the level of the output audio signal is generated as the detected distance is shorter.
Alternatively, in the musical tone control apparatus according to claim 3, as described in claim 9, when the level of the audio signal output from the signal generation unit is equal to or higher than a predetermined threshold, the signal processing unit A configuration may be adopted in which a coefficient for greatly attenuating the level of the output audio signal is generated as the detected distance is shorter.

  In the musical tone control apparatus according to claim 9, as described in claim 10, the signal processing means is detected by the distance detection means when the level of the audio signal output from the signal generation means is equal to or higher than a predetermined threshold value. The threshold may be changed higher as the distance is shorter.

  In the musical tone control device according to claim 1 or 2, until the attack time and / or mute of the rise of the audio signal output from the signal generating means according to the distance detected by the distance detecting means as described in claim 11. The release time may be controlled.

A musical sound control method according to a twelfth aspect of the present invention includes a microphone having signal generation means (corresponding to the microphone element 5 of FIG. 2 in the embodiment) that outputs a sound signal in accordance with an input sound and a sound input target. Step A for detecting the distance from the person's mouth or its surroundings by distance detecting means (corresponding to the distance sensor 6 in FIG. 2 in the embodiment), and from the signal generating means according to the distance detected by Step A Step B for performing effect processing on the audio signal to be output is executed.
In the embodiment, Step A corresponds to the processing of the CPU 10 in FIG. 4, and Step B corresponds to the processing of the CPU 10 and the DSP unit 18 in FIG. 4.

  In a musical sound control method according to a twelfth aspect, as described in the thirteenth aspect, the step B may be configured such that the effect processing is performed by the signal processing means accommodated in the microphone.

  In the musical sound control method according to claim 12 or 13, as described in claim 14, the step A is performed when the reflected light of the light emitted toward the outside of the microphone is received by the distance detecting unit. The distance is detected based on the intensity of the reflected light, and step B corresponds to a coefficient for effect processing according to the distance detected in step A (in the embodiment, a coefficient input to the DSP unit 18 from the CPU 10 in FIG. 4). ) And the audio signal output from the signal generating means may be controlled by the generated coefficient.

15. The musical sound control method according to claim 14, wherein the step B is configured to control the frequency / gain characteristics of the audio signal output from the signal generating means by the generated coefficient, as described in claim 15. Also good.
Alternatively, in the musical sound control method according to claim 14, as described in claim 16, step B is configured to control the cutoff frequency of the audio signal output from the signal generating means by the generated coefficient. May be.
Alternatively, in the musical sound control method according to claim 14, as described in claim 17, step B controls the Q factor which is the steepness of resonance of the audio signal output from the signal generating means by the generated coefficient. You may make it the structure which does.
Alternatively, in the musical sound control method according to claim 14, as described in claim 18, step B is configured to control the output level of the audio signal output from the signal generating means by the generated coefficient. Also good.
Alternatively, in the musical sound control method according to claim 14, as described in claim 18, when the level of the audio signal output from the signal generation unit is equal to or lower than a predetermined threshold, step B is performed by step A. A configuration may be adopted in which a coefficient for greatly attenuating the level of the output audio signal is generated as the detected distance is shorter.
Alternatively, in the musical sound control method according to claim 14, as described in claim 19, when the level of the audio signal output from the signal generating means is equal to or higher than a predetermined threshold, step B is performed by step A. A configuration may be adopted in which a coefficient for greatly attenuating the level of the output audio signal is generated as the detected distance is shorter.

  In the musical tone control method according to claim 20, as described in claim 21, step B is detected by step A when the level of the audio signal output from the signal generating means is equal to or higher than a predetermined threshold value. The threshold may be changed higher as the distance is shorter.

  14. The musical sound control method according to claim 12 or 13, wherein, as described in claim 22, step B is an attack of rising of an audio signal output from the signal generating means according to the distance detected in step A. It may be configured to control the time and / or the release time until the sound is muted.

  According to the musical tone control apparatus and musical tone control method of the present invention, it is possible to obtain an effect that the sound quality is improved and the singing ability is increased more than actual. Furthermore, the effect of suppressing breath noise and unpleasant sounds in the oral cavity can be obtained.

DESCRIPTION OF EMBODIMENTS Hereinafter, a first embodiment to an eighth embodiment of a musical tone control device and a musical tone control method according to the present invention will be described in detail with reference to the drawings.
FIG. 1 is an external view of a karaoke microphone 1 common to each embodiment, and a microphone cover 2 in which a mesh shape or a plurality of holes are formed is attached. FIG. 2 is a diagram showing an internal structure of the microphone 1. An opening 3 is formed in the microphone cover 2, and a substrate 4 is attached inside the microphone 1 behind the microphone cover 2. The substrate 4 has a microphone element 5 that converts sound input through the microphone cover 2 into an electrical signal and outputs a sound signal, a microphone 1 and a voice input target, that is, a singer's mouth or its periphery (for example, lower lip). And a distance sensor 6 for detecting the distance from the nose around the upper lip and the nose around the upper lip or the part under the nose (hereinafter collectively referred to as the mouth). Although not shown in the drawing, the distance sensor 6 is a light emitting element that irradiates infrared rays toward the outside of the microphone 1, receives infrared reflected light reflected from the mouth, photoelectrically converts it, and responds to the amount of reflected light. A light receiving element for outputting a detection signal is incorporated. The audio signal from the microphone element 5 is output from the microphone 1 through the lead wire 7, and the detection signal from the distance sensor 6 is output from the microphone 1 through the lead wire 8.
FIG. 3 is a diagram showing the relationship between the distance D (cm) from the distance sensor 6 to the mouth of the singer and the level E (mV) of the detection signal when the reflected light is photoelectrically converted. As shown in this figure, the distance D (cm) and the detection signal level E (mV) are in an inversely proportional relationship.

  FIG. 4 is a block diagram showing a configuration of a karaoke system common to the embodiments using the musical tone control apparatus according to the present invention. In FIG. 4, a CPU 10 is connected to a program ROM 12, a work RAM 13, an operation switch 14, a display unit 15, a sound source 16, a music data ROM 17, a DSP (Digital Signal Processor) unit 18, and the system bus 11 via the system bus 11. It is connected to an A / D conversion circuit 9 for inputting a detection signal from the distance sensor 6. The CPU 10 controls the entire karaoke system by exchanging data and commands with the above-described units connected via the system bus 11. The A / D conversion circuit 9 converts the level E (mV) of the detection signal from the distance sensor 6 from analog to digital, and based on the relationship shown in FIG. 3, distance data between the microphone 1 and the singer's mouth As input to the CPU 10. Hereinafter, the distance sensor 6 and the A / D conversion circuit 9 are collectively referred to as a “sensor unit”.

  The program ROM 12 stores a musical tone control processing program executed by the CPU 10 and initial data in advance. The work RAM 13 is a work area that temporarily stores data processed by the CPU 10, and is provided with various registers and flags. The operation switch 14 includes a switch group such as a song selection key, a song start key, and a song stop key, and inputs commands and data corresponding to the operation to the CPU 10. The display unit 15 displays a list of karaoke songs and lyrics. The sound source 16 has a built-in waveform ROM for storing PCM waveform data and the like, and generates a digital musical tone signal in accordance with the sound generation command of the CPU 10. The song data ROM 17 stores musical tone data and lyrics data of karaoke accompaniment.

  On the other hand, the microphone element 5 shown in FIG. 2 is connected to the preamplifier 19. The preamplifier 19 amplifies the audio signal output from the microphone element 5 and inputs the amplified audio signal to the A / D conversion circuit 20. The A / D conversion circuit 20 converts the audio signal from analog to digital and inputs it to the DSP unit 18. Hereinafter, the microphone element 5, the preamplifier 19, and the A / D conversion circuit 20 are collectively referred to as a “microphone unit”. The sound source 16 generates an accompaniment musical tone signal based on the waveform data read from the internal waveform ROM in accordance with the karaoke accompaniment read and input from the song data ROM 17 by the CPU 10 to the DSP unit 18. input. The DSP unit 18 performs signal processing on the audio signal input from the microphone unit based on the coefficient from the CPU 10, synthesizes the audio signal and the musical tone signal of the accompaniment music, further performs effect processing, and performs D / A Input to the conversion circuit 21. The D / A conversion circuit 21 converts the composite signal input from the DSP unit 18 from digital to analog, and inputs it to the power amplifier 22 to generate sound from the speaker 23.

  FIG. 5 is a block diagram showing an internal configuration of the DSP unit 18 in the first embodiment. In FIG. 5, an equalizer 181 performs signal processing on the audio signal input from the microphone unit based on the coefficient input from the CPU 10 and inputs the signal to the signal synthesis unit 182. The signal synthesis unit 182 synthesizes the audio signal input from the equalizer 181 and the musical tone signal of the accompaniment input from the sound source 16 in FIG. 4 and inputs the synthesized signal to the total effector 183. The total effector 183 performs final effect processing such as echo, reverb, chorus, etc., on the audio signal synthesized by the signal synthesis unit 182 and the musical tone signal of the accompaniment, and the D / A conversion circuit 21 in FIG. To enter.

Next, the musical tone control processing method of the first embodiment will be described based on the flowchart of the CPU 10 shown in FIGS. 6 and 7 and the characteristic graph shown in FIG.
FIG. 6 is a flowchart of a main routine of the CPU 10 common to the embodiments. First, after predetermined initialization (step SA1), it is determined whether or not the song selection key is turned on (step SA2). When the song selection key is turned on, it is further checked whether or not the song start key is turned on. Discriminate (step SA3). When the song start key is turned on, the selected song data is read from the song data ROM 17 (step SA4) and sent to the sound source 16 (step SA5). Next, DSP control processing is executed (step SA6). After this, it is determined whether or not the song is finished or the stop key is turned on (step SA7). If the song is finished or the stop key is turned on, the process proceeds to step SA2 to detect that the song selection key is turned on. .

  FIG. 7 is a flowchart of the DSP control process in the first embodiment. Distance data is received from the sensor unit (step SB1), and it is determined whether the distance data is 0.5 cm or less, a range from 0.6 cm to 4.9 cm, or 5 cm or more ( Step SB2). When the distance data is 0.5 cm or less, a coefficient for attenuating a frequency of 100 Hz or less of the audio signal by −5 dB is generated (step SB3). If the distance data is in the range from 0.6 cm to 4.9 cm, a coefficient for attenuating a frequency of 100 Hz or less of the audio signal by −10 dB is generated (step SB4). If the distance data is 5 cm or more, a coefficient for attenuating a frequency of 100 Hz or less of the audio signal by −60 dB is generated (step SB5). When a coefficient is generated in any of step SB3, step SB4, and step SB5, the coefficient is sent to the DSP unit 18 (step SB6). Then, the process returns to the main routine of FIG.

  FIG. 8 is a diagram showing a relationship between the frequency f (Hz) of the audio signal and the equalizer gain (EQ · GAIN) by the signal processing in the equalizer 181 of the DSP unit 18 according to the coefficient sent from the CPU 10. In the example of FIG. 8, when the distance data D is 5 cm, 3 cm, and 0.5 cm, the equalizer gain of 100 Hz or less is attenuated as indicated by the dotted line.

As described above, according to the first embodiment, the CPU 10 uses the coefficient generated according to the distance between the microphone 1 and the mouth of the singer who is the target of voice input, and the low frequency range of the voice signal input from the microphone unit. Controls the frequency gain characteristics. Specifically, as shown in FIG. 8, control is performed so that the bass range is greatly attenuated as the mouth approaches the microphone 1. Such effect processing is a mixing technique often performed by professional mixers (arrangers) to improve sound quality.
Therefore, as mixed by a professional mixer, the sound quality can be improved and the singing ability can be increased more than the actual one.

Next, a second embodiment of the present invention will be described. The configuration of the DSP unit 18 in the second embodiment is the same as the configuration of the first embodiment shown in FIG.
FIG. 9 is a flowchart of the DSP control process of the CPU 10 in the second embodiment. Distance data is received from the sensor unit (step SC1), and it is determined whether the distance data is 0.5 cm or less, a range from 0.6 cm to 9.9 cm, or 10 cm or more ( Step SC2). If the distance data is 0.5 cm or less, a coefficient for cutting a frequency of 8 kHz or more of the audio signal is generated (step SC3). If the distance data is in the range from 0.6 cm to 9.9 cm, a coefficient for cutting the frequency of 9 kHz or more of the audio signal is generated (step SC4). If the distance data is 10 cm or more, a coefficient for cutting a frequency of 10 kHz or more of the audio signal is generated (step SC5). When a coefficient is generated in any of step SC3, step SC4, and step SC5, the coefficient is sent to the DSP unit 18 (step SC6). Then, the process returns to the main routine of FIG.

  FIG. 10 is a diagram illustrating a relationship between the frequency f (Hz) of the audio signal and the equalizer gain (EQ · GAIN) by the signal processing in the equalizer 181 of the DSP unit 18 according to the coefficient sent from the CPU 10. In the example of FIG. 10, as shown by the dotted line, when the distance data D is 10 cm, the high frequency of 8 kHz or higher is cut, and when the distance data D is 5 cm, the high frequency of 9 kHz or higher is cut. When the distance data D is 0.5 cm, a high frequency of 10 kHz or higher is cut.

As described above, according to the second embodiment, the CPU 10 uses the coefficient generated according to the distance between the microphone 1 and the mouth of the singer who is the target of voice input, and the high frequency range of the voice signal input from the microphone unit. Control to cut the frequency. Specifically, as shown in FIG. 10, control is performed such that the cut frequency decreases as the mouth approaches the microphone 1.
Therefore, when the singer brings his mouth close to the microphone 1, the high frequency component of the sound is reflected by the hand or the cover 2 of the microphone 1 and added to the high frequency component of the speaker sound in the air to avoid the phenomenon of feeling noisy. By doing so, sound quality can be improved and singing ability can be raised more than actual.

Next, a third embodiment of the present invention will be described. The configuration of the DSP unit 18 in the second embodiment is the same as the configuration of the first embodiment shown in FIG.
FIG. 11 is a flowchart of a DSP control process of the CPU 10 in the third embodiment. Distance data is received from the sensor unit (step SD1), and it is determined whether the distance data is 0.5 cm or less, a range from 0.6 cm to 4.9 cm, or 5 cm or more ( Step SD2). If the distance data is 0.5 cm or less, a coefficient with a Q value of 5 is generated around 200 Hz of the audio signal (step SD3). When the distance data is in the range from 0.6 cm to 4.9 cm, a coefficient with a Q value of 1 is generated around 200 Hz of the audio signal (step SD4). If the distance data is 5 cm or more, a coefficient is generated so that the Q value is 0.5 centering on 200 Hz of the audio signal (step SD5). When a coefficient is generated in any of step SD3, step SD4, and step SD5, the coefficient is sent to the DSP unit 18 (step SD6). Then, the process returns to the main routine of FIG.

  FIG. 12 is a diagram showing gain characteristics of the equalizer 181 in which the value of Q (Q factor) is changed by a coefficient generated according to the distance D (cm) between the microphone 1 and the singer's mouth. The equalizer 181 is a parametric equalizer that can freely design a Q factor having sharp characteristics. The equalizer 181 increases the Q factor and cuts the frequency band centered on 200 Hz as the singer's mouth is closer to the microphone 1 based on the coefficient sent from the CPU 10.

As described above, according to the third embodiment, the CPU 10 uses the coefficient generated according to the distance between the microphone 1 and the mouth of the singer who is the target of voice input to calculate 200 Hz of the voice signal input from the microphone unit. By cutting the center frequency band and preventing it from becoming unpleasant in hearing, the sound quality can be improved and the singing ability can be increased more than the actual one.
Furthermore, since the equalizer in the second embodiment and the equalizer in the third embodiment can be configured by the same parametric equalizer, the processing load of the DSP unit 18 can be reduced and the resource utilization rate can be increased.

Next, a fourth embodiment of the present invention will be described.
FIG. 13 is a block diagram showing an internal configuration of the DSP unit 18 in the fourth embodiment. In FIG. 13, a VCA (Voltage Controlled Amplifier) 184 normally has a one-to-one amplification factor (0 dB) in input / output characteristics, but the amplification factor depends on the DC voltage coefficient sent from the CPU 10. Changes. Therefore, the level of the audio signal input from the microphone unit is changed according to the coefficient generated according to the distance between the microphone 1 and the mouth of the singer who is the target of audio input.

  FIG. 14 is a flowchart of a DSP control process of the CPU 10 in the fourth embodiment. Distance data is received from the sensor unit (step SE1), and it is determined whether or not the distance data is 5 cm or less (step SE2). If it is 5 cm or less, it is further determined whether it is 3 cm or less (step SE3). If it is 3 cm or less, it is further determined whether it is 0.5 cm or less (step SE4). If it is 0.5 cm or less, a coefficient for multiplying the input level from the microphone unit by 1/3 is generated (step SE5). In step SE4, if the distance data is in the range from 0.5 cm to 3 cm, a coefficient for halving the input level from the microphone unit is generated (step SE6). If the distance data is in the range from 3 cm to 5 cm in step SE3, a coefficient for multiplying the input level from the microphone unit by 1 / 1.5 is generated (step SE7). In step SE2, if the distance data is larger than 5 cm, a coefficient for outputting (1 times) the input level from the microphone unit as it is is generated (step SE8). After generating a coefficient of any value, the coefficient is sent to the DSP unit 18 (step SE9). Then, the process returns to the main routine of FIG.

  FIG. 15 is a diagram illustrating the input / output characteristics of the VCA 184 that controls the level of the audio signal input from the microphone unit using a coefficient generated according to the distance D (cm) between the microphone 1 and the singer's mouth. As shown in FIG. 15, the closer the distance between the microphone 1 and the singer's mouth is, the smaller the amplification factor with respect to the input level of the audio signal to the VCA 184 is, and the lower the output level is.

  As described above, according to the fourth embodiment, the CPU 10 determines the level of the audio signal input from the microphone unit based on the coefficient generated according to the distance between the microphone 1 and the mouth of the singer who is the target of audio input. Even when controlling and singing with the mouth close to the microphone 1, by preventing breath noise and unpleasant sound in the oral cavity, the sound quality can be improved and the singing ability can be enhanced more than the actual.

Next, a fifth embodiment of the present invention will be described.
FIG. 16 is a block diagram showing an internal configuration of the DSP unit 18 in the fifth embodiment. In FIG. 16, the expander 185 normally has a one-to-one gain input / output characteristic (amplification factor = 0 dB), but controls a small audio signal input from the microphone unit to be smaller. However, unlike a normal expander, control for increasing a large audio signal input from the microphone unit is not performed. Other configurations of the DSP unit 18 are the same as those in the first embodiment.

  FIG. 17 is a flowchart of a DSP control process of the CPU 10 in the fifth embodiment. Distance data is received from the sensor unit (step SF1), and it is determined whether or not the input level of the audio signal from the microphone unit is −20 dB or less (step SF2). If the input level is −20 dB or less, it is determined whether or not the distance data from the sensor unit is smaller than 5 cm (step SF3). If it is smaller than 5 cm, it is further determined whether it is smaller than 3 cm (step SF4). If it is smaller than 3 cm, it is further determined whether it is smaller than 0.5 cm (step SF5).

  When the distance data from the sensor unit is smaller than 0.5 cm, a coefficient for increasing the input level from the microphone unit by 1/3 is generated (step SF6). In step SF5, when the distance data from the sensor unit is within a range of 0.5 cm or more and smaller than 3 cm, a coefficient for halving the input level from the microphone unit is generated (step SF7). In step SF4, when the distance data from the sensor unit is in a range of 3 cm or more and smaller than 5 cm, a coefficient for increasing the input level from the microphone unit by a factor of 1 / 1.5 is generated (step SF8). If the distance data from the sensor unit is 5 cm or more in step SF3, or if the input level of the audio signal from the microphone unit is greater than −20 dB in step SF2, the input from the microphone unit is output as it is (1 A coefficient to be multiplied is generated (step SF9). When a coefficient having any value is generated, the coefficient is sent to the DSP unit (step SF10). Then, the process returns to the main routine of FIG.

  FIG. 18 is a diagram illustrating input / output characteristics of an expander 185 that controls the level of an audio signal input from the microphone unit based on a coefficient generated according to the distance D (cm) between the microphone 1 and the singer's mouth. . Normally, the input / output characteristics of the expander 185 are 1: 1, but as shown in FIG. 18, when the output level of the audio signal from the microphone is -20 dB or less, the microphone 1 and the singer As the distance from the mouth is closer, the level of the audio signal input from the microphone unit is greatly attenuated.

  As described above, according to the fifth embodiment, the CPU 10 determines the level of the audio signal input from the microphone unit by the coefficient generated according to the distance between the microphone 1 and the mouth of the singer who is the target of audio input. When the output level of the audio signal is below a predetermined value, it is noticeable when the output level of the audio signal is low, such as breath noise, sound due to saliva viscosity in the oral cavity, or sound of the tongue in the oral cavity. By preventing the unpleasant sound which becomes, sound quality can be improved and singing power can be improved more than actual.

Next, a sixth embodiment of the present invention will be described.
FIG. 19 is a block diagram showing an internal configuration of the DSP unit 18 in the sixth embodiment. In FIG. 19, a compressor 186 controls to compress a large audio signal input from the microphone unit. Other configurations of the DSP unit 18 are the same as those in the first embodiment.

  FIG. 20 is a flowchart of the DSP control processing of the CPU 10 in the sixth embodiment. Distance data is received from the sensor unit (step SG1), and it is determined whether or not the input level from the microphone unit is -5 dB or more (step SG2). When the input level is −5 dB or more, it is determined whether or not the distance data from the sensor unit is smaller than 5 cm (step SG3). If it is smaller than 5 cm, it is further determined whether it is smaller than 3 cm (step SG4). If it is smaller than 3 cm, it is further determined whether it is smaller than 0.5 cm (step SG5).

  If the distance data from the sensor unit is smaller than 0.5 cm, a coefficient for increasing the input level from the microphone unit by 1/3 is generated (step SG6). In step SG5, if the distance data from the sensor unit is in the range of 0.5 cm or more and smaller than 3 cm, a coefficient for halving the input level from the microphone unit is generated (step SG7). In step SG4, if the distance data from the sensor unit is in the range of 3 cm or more and less than 5 cm, a coefficient for increasing the input level from the microphone unit by 1 / 1.5 times is generated (step SG8). If the distance data from the sensor unit is 5 cm or more in step SG3, or if the input level of the audio signal from the microphone unit is less than −5 dB in step SG2, the input from the microphone unit is output as it is (1 A coefficient to be multiplied is generated (step SG9). When a coefficient having any value is generated, the coefficient is sent to the DSP unit (step SG10). Then, the process returns to the main routine of FIG.

  FIG. 21 shows the input / output characteristics of the compressor 186 of the sixth embodiment that controls the level of the audio signal input from the microphone unit by the coefficient generated according to the distance D (cm) between the microphone 1 and the singer's mouth. FIG. Normally, the input / output characteristic of the compressor 186 is 1: 1 (compression ratio = 0 dB). However, as shown in FIG. 21, when the output level of the audio signal from the microphone unit is −5 dB or more, the microphone 1 As the distance from the singer's mouth gets closer, the output level of the audio signal is greatly attenuated.

  As described above, according to the sixth embodiment, the CPU 10 determines the level of the audio signal input from the microphone unit by the coefficient generated according to the distance between the microphone 1 and the mouth of the singer who is the target of audio input. If the level of the audio signal is controlled to be equal to or higher than a predetermined value, when the distance between the microphone 1 and the mouth of the singer becomes close, the input level to the power amplifier 22 is too high and sound distortion occurs. Can be improved, and the sound quality can be improved and the singing ability can be increased more than actual.

Next, a seventh embodiment of the present invention will be described. The configuration of the DSP unit 18 in the seventh embodiment is the same as the configuration of the sixth embodiment shown in FIG.
FIG. 22 is a flowchart of the DSP control process of the CPU 10 in the seventh embodiment. Distance data is received from the sensor unit (step SH1), and it is determined whether or not the distance data from the sensor unit is 5 cm or less (step SH2). If it is 5 cm or less, it is further determined whether or not it is 3 cm or less (step SH3). If it is 3 cm or less, it is further determined whether or not it is smaller than 0.5 cm (step SH4). If it is smaller than 0.5 cm, −5 dB is set in the threshold register TH (step SH5). If the distance is not less than 0.5 cm and not more than 3 cm in step SH4, TH is set to −10 dB (step SH6). If the distance is larger than 3 cm and not larger than 5 cm in step SH3, TH is set to −20 dB (step SH7). If the distance is greater than 5 cm in step SH2, TH is set to -40 dB (step SH8).

  In step SH5, step SH6, step SH7, or step SH8, after generating a threshold value according to the distance data from the sensor unit, it is determined whether the input level from the microphone unit is greater than or less than the TH threshold value. It discriminate | determines (step SH9). If the input level from the microphone unit is larger than the threshold value of TH, a coefficient for outputting the input from the microphone unit as it is is generated (step SH10). On the other hand, when the input level from the microphone unit is equal to or less than the threshold value of TH, a coefficient to be output by multiplying the input from the microphone unit by 1/4 is generated (step SH11). Next, the coefficient generated in step SH10 or SH11 is sent to the DSP unit 18 (step SH12). Then, the process returns to the main routine of FIG.

  FIG. 23 shows the input / output characteristics of the compressor 186 of the seventh embodiment that controls the level of an audio signal input from the microphone unit by a coefficient generated according to the distance D (cm) between the microphone 1 and the singer's mouth. FIG. The input / output characteristics of the compressor 186 are one-to-one (compression rate = 0 dB) when the level of the audio signal input from the microphone unit is less than or equal to the threshold value. However, when the level of the audio signal becomes larger than the threshold value, the compressor 186 The output level from is attenuated. In this case, as shown in FIG. 23, the threshold value to be set increases as the distance between the microphone 1 and the singer's mouth becomes shorter.

  As described above, according to the seventh embodiment, the CPU 10 determines the level of the audio signal input from the microphone unit based on the coefficient generated according to the distance between the microphone 1 and the mouth of the singer who is the target of audio input. When the distance between the microphone 1 and the singer's mouth is increased, the input level to the power amplifier 22 is prevented from being too low and the sound from the speaker becomes poor, so that the sound quality is improved and the singing is improved. You can increase your power.

Next, an eighth embodiment of the present invention will be described. In the eighth embodiment, the DSP unit 18 controls the envelope of the audio signal by the ADSR (Attack, Decay, Sustain, Release) method. Since the ADSR system is a well-known technique, description thereof is omitted.
24 and 25 are flowcharts of the DSP control processing of the CPU 10 in the eighth embodiment. In the flowchart of FIG. 24, the level of the new audio signal of this time input from the microphone unit is stored in the register LEVEL (step SJ1), the LEVEL value is -5 dB or more, and the level of the previous audio signal is stored. It is determined whether or not the value of the registered register LEVEL is smaller than -5 dB (step SJ2). That is, it is determined whether or not the level of the audio signal that was smaller than the previous reference level (−5 dB) has increased this time and has become equal to or higher than the reference level. When the level of the current audio signal becomes -5 dB or higher, the attack flag ATF is set to 1 and the release flag REF is reset to 0 because the voice is rising (step SJ3).

  In step SJ2, if the level of the current audio signal is lower than −5 dB, it is determined whether or not the level of the previous audio signal stored in LEVEL is −5 dB or higher (step SJ4). That is, it is determined whether or not the level of the audio signal that is equal to or higher than the previous reference level has decreased this time and becomes lower than the reference level. When the level of the current audio signal is less than -5 dB, the mute has started, so the attack flag ATF is reset to 0 and the release flag REF is set to 1 (step SJ5).

  After ATF and REF are set to 1 or 0, distance data is received from the sensor unit (step SJ6), and it is determined whether or not ATF is 1 (step SJ7). If ATF is 1, the attack time corresponding to the distance data is stored in the register ATT (step SJ8). On the other hand, if ATF is 0, it is determined whether or not REF is 1 (step SJ9). If REF is 1, the release time corresponding to the distance data is stored in the register RET (step SJ9). Step SJ10).

  After the attack time is stored in ATT in step SJ8, or after the release time is stored in RET in step SJ10, start flag STF is set to 1 (step SJ11). After setting STF to 1, or when ATF is 0 in step SJ7 and REF is 0 in step SJ9, or the change between the level of the previous audio signal and the level of the current audio signal is −5 dB. If it does not cross the reference level, the value stored in LEVEL is transferred to LEVEL (step SJ12).

  Next, in the flowchart of FIG. 25, it is determined whether or not the STF is 1 (step SJ13). If the STF is 1, the minimum clock time (for example, 96th notes) of the progression of the music is obtained. It is determined whether or not (time) has elapsed (step SJ14). When STF is 0, or when the minimum clock time has not elapsed, the process returns to the main routine. On the other hand, when the minimum clock time has elapsed, it is determined whether or not ATF is 1 (step SJ15). If ATF is 1, the value of ATT is decremented (step SJ16). Thereafter, it is determined whether or not the value of ATT has reached 0 (step SJ17). If the ATT value does not reach 0, the process returns to the main routine. However, when the ATT value reaches 0, in order to shift from attack to decay, a coefficient for reducing the input from the microphone unit to 1/4 is set. Generate (step SJ18). Then, ATF is reset to 0 (step SJ19).

  In step SJ15, if ATF is 0, it is determined whether REF is 1 (step SJ20). If REF is 1, the value of RET is decremented (step SJ21). Thereafter, it is determined whether or not the value of RET has reached 0 (step SJ22). When the value of RET does not reach 0, the process returns to the main routine. However, when the value of RET reaches 0, since the sound is muted, a coefficient for outputting the input from the microphone unit (level = 0) as it is is obtained. Generate (step SJ23). Then, REF is reset to 0 (step SJ24).

  After resetting the ATF in step SJ19 or after resetting REF in step SJ24, the STF is reset to 0 (step SJ25). Then, the coefficient generated in step SJ18 or step SJ23 is sent to the DSP unit 18 (step SJ26). After this, or when ATF is 0 in step SJ15 and REF is 0 in step SJ20, the process returns to the main routine of FIG.

FIG. 26 is a diagram illustrating characteristics of the DSP unit 18 that changes the release time according to the distance D (cm) between the microphone 1 and the mouth of the singer. According to this characteristic, the shorter the distance between the microphone 1 and the singer's mouth, the shorter the release time.
FIG. 27 is a diagram illustrating characteristics of the DSP unit 18 that changes the release time according to the distance D (cm) between the microphone 1 and the mouth of the singer. According to this characteristic, the shorter the distance between the microphone 1 and the singer's mouth, the shorter the attack time.

As described above, according to the eighth embodiment, the CPU 10 determines the release time and the attack time of the audio signal input from the microphone unit according to the distance between the microphone 1 and the mouth of the singer who is the target of audio input. Control and improve the sound quality by enhancing the sound quality by emphasizing the singing voice by applying the effect of shortening the release time and making the tongue clearer as the distance between the microphone 1 and the singer's mouth is shorter Can be higher than actual.
On the other hand, when the distance between the microphone 1 and the mouth of the singer is close, the voice tends to be sharper. Therefore, by shortening the attack time and rounding the voice earlier, the sound quality is improved and the singing ability is actually increased. Can also be increased.

As described above, according to the first to eighth embodiments, the microphone element 5 that outputs a sound signal according to the sound input to the microphone 1, and the singer who is the target person of the microphone 1 and the sound input. A distance sensor 6 that detects a distance D (cm) from the mouth of the computer, and a CPU 10 and a DSP unit 18 that perform effect processing on the audio signal output from the microphone element 5 according to the distance detected by the distance sensor 6 And.
Therefore, the effect of improving the sound quality and enhancing the singing ability than the actual one can be obtained. Furthermore, in 4th and 5th embodiment, the effect of suppressing breath noise and the unpleasant sound in an oral cavity is acquired.

In each of the above embodiments, the distance sensor 6 emits light toward the outside of the microphone 1 and detects the distance between the singer's mouth and the microphone 1 based on the intensity of the reflected light. The coefficient of the effect processing is generated according to the distance detected by 6, the generated coefficient is sent to the DSP unit 18, and the audio signal output from the microphone 1 is controlled.
Therefore, it is possible to detect the distance at high speed by using light, and it is possible to perform an effect process that can respond sufficiently even in the case of a song with a fast tempo.

As a modification of each of the above embodiments, a signal processing unit including the DSP unit 18 and the CPU 10 may be provided inside the microphone 1. For example, an LSI incorporating a DSP, a CPU, and an electronic circuit related to signal processing is mounted on the substrate 4 of FIG.
According to the configuration of this modified example, since the audio signal from the microphone element is signal-processed inside the microphone, the S / N ratio and response characteristics can be improved.

The external view of the microphone in embodiment of this invention. The figure which shows the internal structure of the microphone of FIG. The figure showing the relationship between the distance from a distance sensor to a singer's mouth, and the level of a detection signal. The block diagram which shows the structure of a karaoke system common to each embodiment to which the musical tone control apparatus of this invention is applied. The block diagram which shows the internal structure of the DSP part in 1st Embodiment. The flowchart of the main routine of CPU common to each embodiment. The flowchart of the DSP control processing in 1st Embodiment. The figure which shows the relationship between the frequency of the audio | voice signal in 1st Embodiment, and an equalizer gain. The flowchart of DSP control processing of CPU in 2nd Embodiment. The figure which shows the relationship between the frequency of the audio | voice signal and equalizer gain in 2nd Embodiment. The flowchart of DSP control processing of CPU in 3rd Embodiment. The figure which shows the gain characteristic of the equalizer which changed the value of Q in 3rd Embodiment. The block diagram which shows the internal structure of the DSP part in 4th Embodiment. The flowchart of DSP control processing of CPU in 4th Embodiment. The figure which shows the input / output characteristic of VCA of the DSP part in 4th Embodiment. The block diagram which shows the internal structure of the DSP part in 5th Embodiment. The flowchart of DSP control processing of CPU in 5th Embodiment. The figure which shows the input-output characteristic of the expander of the DSP part in 5th Embodiment. The block diagram which shows the internal structure of the DSP part in 6th Embodiment. The flowchart of DSP control processing of CPU in 6th Embodiment. The figure which shows the input-output characteristic of the compressor of the DSP part in 6th Embodiment. The flowchart of DSP control processing of CPU in 7th Embodiment. The figure which shows the input-output characteristic of the compressor of the DSP part in 7th Embodiment. The flowchart of the DSP control processing of CPU in 8th Embodiment. The flowchart of the DSP control process following FIG. The figure which shows the characteristic of the DSP part which changes the release time in 8th Embodiment. The figure which shows the characteristic of the DSP part which changes the attack time in 8th Embodiment.

Explanation of symbols

DESCRIPTION OF SYMBOLS 1 Microphone 2 Microphone cover 3 Opening part 4 Board | substrate 5 Microphone element 6 Distance sensor 9 A / D conversion circuit 10 CPU
12 Program ROM
13 Work RAM
16 sound source 17 song data ROM
18 DSP unit 181 Equalizer 182 Synthesis circuit 183 Total effector 184 VCA
185 expander

Claims (22)

Signal generating means for outputting an audio signal in accordance with audio input to the microphone;
A distance detecting means for detecting a distance between the microphone and the mouth of the subject of voice input or the vicinity thereof;
Signal processing means for effecting the sound signal output from the signal generating means according to the distance detected by the distance detecting means;
Musical sound control device with   The musical tone control apparatus according to claim 1, wherein the signal processing means is accommodated in the microphone.   The distance detecting means irradiates light outside the microphone and detects the distance based on the intensity of the reflected light, and the signal processing means performs effect processing according to the distance detected by the distance detecting means. The musical tone control apparatus according to claim 1 or 2, wherein a coefficient is generated, and an audio signal output from the signal generating means is controlled by the generated coefficient.   4. The musical tone control apparatus according to claim 3, wherein the signal processing means controls a frequency / gain characteristic of an audio signal output from the signal generating means according to the generated coefficient.   4. The musical tone control apparatus according to claim 3, wherein the signal processing means controls a cut-off frequency of an audio signal output from the signal generating means according to the generated coefficient.   4. The musical tone control apparatus according to claim 3, wherein the signal processing unit controls a Q factor which is a resonance steepness of an audio signal output from the signal generating unit according to the generated coefficient.   4. The musical tone control apparatus according to claim 3, wherein the signal processing means controls the output level of the audio signal output from the signal generating means according to the generated coefficient.   The signal processing unit increases the level of the output audio signal as the distance detected by the distance detection unit is closer when the level of the audio signal output from the signal generation unit is equal to or less than a predetermined threshold. 4. The musical tone control apparatus according to claim 3, wherein a coefficient for attenuation is generated.   The signal processing unit increases the level of the output audio signal as the distance detected by the distance detection unit is closer when the level of the audio signal output from the signal generation unit is equal to or greater than a predetermined threshold. 4. The musical tone control apparatus according to claim 3, wherein a coefficient for attenuation is generated.   The signal processing means, when the level of the audio signal output from the signal generation means is equal to or higher than the predetermined threshold, changes the threshold higher as the distance detected by the distance detection means is closer. The musical tone control apparatus according to claim 9.   The signal processing means controls the attack time of the rising edge of the audio signal output from the signal generating means and / or the release time until mute according to the distance detected by the distance detecting means. Item 3. The musical tone control device according to Item 1 or 2. A step A for detecting a distance between a microphone having a signal generating means for outputting a sound signal in accordance with an input sound and a mouth of the subject of sound input or the vicinity thereof by a distance detecting means;
Applying effect processing to the audio signal output from the signal generating means according to the distance detected in step A;
Perform a musical sound control method.   13. The musical tone control method according to claim 12, wherein the step B performs effect processing by signal processing means accommodated in the microphone.   The step A detects the distance by the intensity of the reflected light when the reflected light of the light emitted toward the outside of the microphone is received by the distance detecting means, and the step B is performed by the step A 14. The musical tone control method according to claim 12, wherein a coefficient for effect processing is generated according to the detected distance, and an audio signal output from the signal generating means is controlled based on the generated coefficient.   15. The musical tone control method according to claim 14, wherein the step B controls a frequency / gain characteristic of an audio signal output from the signal generating means according to the generated coefficient.   15. The musical sound control method according to claim 14, wherein the step B controls a cut-off frequency of the audio signal output from the signal generating means according to the generated coefficient.   15. The musical tone control method according to claim 14, wherein the step B controls a Q factor which is a steepness of resonance of an audio signal output from the signal generating means according to the generated coefficient.   15. The musical sound control method according to claim 14, wherein the step B controls the output level of the audio signal output from the signal generating means according to the generated coefficient.   In the step B, when the level of the audio signal output from the signal generating means is not more than a predetermined threshold value, the level of the output audio signal is greatly attenuated as the distance detected in the step A is closer. 15. The musical sound control method according to claim 14, wherein a coefficient for generating the sound is generated.   In the step B, when the level of the audio signal output from the signal generating means is equal to or greater than a predetermined threshold, the level of the output audio signal is greatly attenuated as the distance detected in the step A is closer. 15. The musical sound control method according to claim 14, wherein a coefficient for generating the sound is generated.   In the step B, when the level of the audio signal output from the signal generating means is equal to or higher than the predetermined threshold, the threshold is changed higher as the distance detected in the step A is closer. The musical tone control method according to claim 20.   13. The step B controls the attack time of the rising edge of the audio signal output from the signal generating means and / or the release time until mute according to the distance detected in the step A. Or the musical sound control method of 13.

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