Voice processing device

Abstract

To provide a voice processing device for enabling a hearing-impaired person to easily hear voice without using an auxiliary device by allowing an utterer to use the auxiliary device.SOLUTION: A portable voice processing device (1) includes: a voice processing unit (10) for executing voice processing to clarify voice with respect to a voice signal supplied from a microphone (21) connected to a support member (24) by which the microphone is positioned close to the mouth of a user; a speaker (22) for converting the voice signal after the execution of the voice processing into voice; and a housing for storing the voice processing unit (10).SELECTED DRAWING: Figure 2

Description

[Technical field]

[0001] The present invention relates to an audio processing device. [Background technology]

[0002] 2. Description of the Related Art The proportion of the population suffering from hearing impairment has been increasing due to aging, and assistive devices such as hearing aids and sound amplifiers have become common. [Prior art documents] [Patent documents]

[0003] [Patent Document 1] JP 2016-110050 A Summary of the Invention [Problem to be solved by the invention]

[0004] Fig. 5 of Patent Document 1 discloses an audio processing device for making audio coming from speakers of audio equipment, broadcasting equipment, etc. easier to hear clearly. This audio processing device can be read as the audio processor in this specification. Therefore, hereinafter, the audio processing device described in Patent Document 1 will be referred to as the audio processor.

[0005] However, the voice processing device described in Patent Document 1 is mainly targeted at audio equipment and broadcasting facilities installed in places where many people gather, such as buildings, public buildings, shopping centers, sports facilities, etc. Therefore, this voice processing device emits voice-processed voice through speakers that are installed in advance in places where many people gather. Therefore, this voice processing device cannot be carried around, and therefore cannot be used in a state where the speaker and the voice processing device can move together.

[0006] One aspect of the present invention has been made in consideration of the above-mentioned problems, and an object of the present invention is to provide a voice processing device that can be used in a mobile state together with a speaker. [Means for solving the problem]

[0007] In order to solve the above problems, an information processing device according to a first aspect of the present invention is a portable audio processing device, which includes an audio processor that performs audio processing on an audio signal supplied from a microphone to clarify the audio, a speaker that converts the processed audio signal into audio, and a housing that accommodates at least the audio processor.

[0008] According to the above configuration, since the voice processing device is configured to be portable, it is possible to provide a voice processing device that can be used in a mobile state together with a speaker.

[0009] Since the speaker and the voice processing device can be moved together, the voice processing device can be used in various situations, such as when a store clerk interacts with a customer, when a staff member at an event venue guides spectators, when a local government official provides information to citizens at a disaster site, etc. Therefore, even if a hearing-impaired person is included among the above-mentioned customers, spectators, citizens, etc., it becomes easy for the speaker to convey the message he or she wants to convey to the hearing-impaired person.

[0010] In these cases, the person to whom the speaker speaks using the processed voice may be a specific person or people facing the speaker, or it may be an unspecified person or people, such as people passing by on the street.

[0011] In addition, in an audio processing device according to a second aspect of the present invention, in addition to the configuration of the audio processing device according to the first aspect described above, a configuration is adopted in which the housing accommodates the speaker in addition to the audio processor.

[0012] According to the above configuration, the voice processor and the speaker are integrally housed in a single housing, making it easy for the speaker to carry the voice processor.

[0013] In addition, in an audio processing device according to a third aspect of the present invention, in addition to the configuration of the audio processing device according to the second aspect described above, a microphone that converts audio into an audio signal is further provided, and the housing accommodates the microphone in addition to the audio processor and the speaker.

[0014] According to the above configuration, the voice processor, the speaker, and the microphone are integrally housed in a single housing, making it easy for the speaker to carry the voice processor with them. An example of such a voice processor is a device called an electronic megaphone or loudspeaker.

[0015] In addition, in the audio processing device according to the fourth aspect of the present invention, in addition to the configuration of the audio processing device according to the first or second aspect described above, a configuration is adopted which further includes a microphone that converts audio into an audio signal.

[0016] The information processing device according to one aspect of the present invention can also be sold as a package including a microphone. Effect of the Invention

[0017] According to one aspect of the present invention, it is possible to provide a voice processing device that can be used in a mobile state together with a speaker. [Brief description of the drawings]

[0018] [Figure 1] 1 is a block diagram of a voice processing device according to a first embodiment of the present invention. [Diagram 2] The left diagram is an example of an external view of the voice processing device shown in FIG. 1, and the right diagram is a schematic diagram of a user wearing the voice processing device shown in FIG. [Diagram 3] 2 is a graph showing frequency characteristics in the audio processing device shown in FIG. [Figure 4] FIG. 11 is a block diagram showing an example of a configuration of a voice processing device according to a second embodiment of the present invention. [Diagram 5] FIG. 2 is a diagram illustrating formant components contained in a speech signal. [Figure 6] 5 is a graph showing adjustment characteristics of an input buffer section included in the audio processing device of FIG. 4. [Figure 7] 5 is a circuit diagram showing an example of a specific configuration of a filter and a correction signal generating unit included in the audio processing device of FIG. 4. [Figure 8] 5 is a graph showing a phase correction characteristic of a filter included in the audio processing device of FIG. [Figure 9] 11 is a graph illustrating a case where a first audio signal and a second audio signal are combined without performing phase correction. [Figure 10] 11 is a graph showing frequency characteristics of a synthetic speech signal when phase correction is not performed. [Figure 11] 5 is a graph showing an example of adjustment of an audio signal by an adjustment unit included in the audio processing device of FIG. 4. [Figure 12] 6 is a graph showing changes over time in phase and output of an audio signal, a first audio signal before and after correction, and a synthetic audio signal. [Figure 13] 13 is a graph showing the overall output characteristic of a synthesis unit. DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS

[0019] [Embodiment 1] Hereinafter, one embodiment of the present invention will be described in detail.

[0020] <Outline of voice processing device> In recent years, with the aging of the population, it has been reported that there are frequent cases where important information spoken by a speaker cannot be conveyed to a listener who has hearing loss. The general idea is to simply increase the speaker's voice volume, but hearing loss can be classified into conductive hearing loss and sensorineural hearing loss caused by damage to the inner ear. In particular, sensorineural hearing loss makes it impossible to understand language even at high volumes, and it has been elucidated that the cause of this is a decrease in the function of the cochlea, which is the organ of hearing, to convert high frequencies into auditory signals.

[0021] In situations where hearing function has declined due to aging or other factors, communication through conversation in daily life becomes sparse, so hearing-impaired people generally use hearing aids, sound amplifiers, and other assistive devices. These assistive devices increase the volume and / or process the frequency of sounds picked up by a microphone to make them easier to hear, and then output the sounds from earphones.

[0022] However, in places such as nursing homes and care facilities where there are a large number of hearing-impaired people, it is inefficient and expensive to require all of the hearing-impaired people to wear an assistive device.

[0023] The inventor therefore sought a method for enabling a speaker to use an auxiliary device to make it easier for a hearing-impaired person to hear speech without using the auxiliary device.

[0024] A voice processing device 1 according to one embodiment of the present invention will be described below with reference to Figs. 1 to 3. Fig. 1 is a block diagram of the voice processing device 1. The left diagram of Fig. 2 is an example of an external view of the voice processing device 1, and the right diagram of Fig. 2 is a schematic diagram of a user wearing the voice processing device 1. Fig. 3 is a graph showing frequency characteristics in the voice processing device 1.

[0025] <Idea for voice processing device> The voice processing device 1, which is a type of assistive device, is equipped with technology that enables difficult-to-hear languages ​​to be accurately understood by auxiliary correction of frequency components that play an important role in understanding language, and achieves this using a method that is completely opposite to the conventional idea of ​​correcting voice signals on the speaker's side.

[0026] The speech processing device 1 has a function that enables the speaker to arbitrarily set the strength of signal processing components based on the listener's language comprehension ability, thereby enabling a comfortable conversation to be realized.

[0027] <Configuration of voice processing device> 1, the audio processing device 1 includes a microphone 21, an audio processor 10, a speaker 22, and a housing. In this embodiment, the housing houses at least the audio processor 10.

[0028] The voice processing device 1 is configured to be portable. The voice processing device 1 may be configured to be wearable on a part of the body or clothing of a speaker, or may be configured to be provided with a holder that can be held by the speaker.

[0029] In this embodiment, the voice processing device 1 includes a clip provided on the back surface of the housing (the surface facing the speaker's body), although this is not shown in the left diagram of FIG. 2. The clip is an example of a configuration for attaching the voice processing device 1 to the trousers, skirt, or belt worn by the speaker. Therefore, the voice processing device 1 is configured so that the speaker can carry it while wearing it on his or her waist (see the right diagram of FIG. 2). Note that the configuration that can be attached to a part of the speaker's body or clothing is not limited to the above-mentioned clip, and may be, for example, a belt or a strap. The voice processing device 1 may also be a wristwatch type. The wristwatch type voice processing device 1 is also an example of a wearable device.

[0030] An example of the voice processing device 1 having a holding part that can be held by a speaker is an electronic megaphone equipped with the voice processor 10. A grip part of the electronic megaphone that is held by the speaker is an example of a holding part that can be held by the speaker.

[0031] The voice processing device 1 is configured to weigh 10 kg or less so that it can be carried by an average adult. The lighter the weight of the voice processing device 1, the better. In addition, when the voice processing device 1 is configured to be wearable on the speaker's body or clothing, the weight of the voice processing device 1 is preferably 1 kg or less.

[0032] (microphone) The microphone 21 converts the voice uttered by the speaker into a voice signal SV and supplies the voice signal SV to the port P11 of the voice processor 10 to which the microphone 21 is connected. In this embodiment, the voice signal SV is an analog signal and an electrical signal. However, the microphone 21 may be configured to convert the voice uttered by the user on the transmitting side into the voice signal SV, which is a digital signal.

[0033] For example, an electric condenser microphone can be used as the microphone 21. A direct current voltage for driving is applied to the electric condenser microphone. In addition, electric condenser microphones are often connected using connectors of different types depending on the manufacturer or model.

[0034] (audio processor) The audio processor according to one embodiment of the present invention is configured to perform audio processing on an audio signal supplied from a microphone 21 to clarify the audio, and to output the audio signal that has been subjected to the audio processing. In the following, an audio processor 10 that is an example of an audio processor according to one embodiment of the present invention will be described.

[0035] As shown in Fig. 1, the audio processor 10 includes a port P11, a port P12, a branching unit 11, a phase corrector 12, a filter (band-pass filter) 13, an amplifier 14, an adjuster 15, and an additive synthesizer 16. The audio processor 10 is provided so as to be interposed between a microphone 21 and a speaker. The branching unit 11, the filter 13, the amplifier 14, and the additive synthesizer 16 are examples of an audio processing unit. The audio processor 10 further includes a power supply unit, a detection unit, and a control unit, which are not shown in Fig. 1.

[0036] The port P11 is a port connected to the microphone 21 and is an example of a first port. The port P11 and the microphone 21 are connected using a cable or wiring. A voice signal SV converted from a voice uttered by a speaker by the microphone 21 is supplied from the microphone 21 to a terminal of the port P11.

[0037] The speech signal SV includes a first formant component, a second formant component, ..., and an nth formant component, as shown in Fig. 1 of Patent Document 1. Note that n depends on the user on the transmitting side and is a positive integer of at least 4, although there may be some individual differences. Note that the first formant component, the second formant component, ..., and the nth formant component can be read as the first formant, the second formant, ..., and the nth formant, respectively, shown in Fig. 1 of Patent Document 1.

[0038] It should be noted that the first formant component, the second formant component, . . . , and the nth formant component are described in Patent Document 1, and therefore a description thereof will be omitted in this embodiment.

[0039] The branching unit 11 branches the audio signal SV into a first audio signal SV1 and a second audio signal SV2. In this embodiment, the branching unit 11 is configured so that the intensity ratio between the first audio signal SV1 and the second audio signal SV2 is 1:1, that is, the distribution ratio is 1:1. However, the distribution ratio of the branching unit 11 is not limited to 1:1 and can be set appropriately.

[0040] The spectrum of each of the first and second audio signals SV1 and SV2 is similar to the spectrum of the audio signal SV. That is, each of the first and second audio signals SV1 and SV2 includes a first formant component, a second formant component, ..., an nth formant component.

[0041] The filter 13 is a high-pass filter that removes low-frequency components, which are components with frequencies less than 400 Hz, from the first audio signal SV1 that has passed through the branching unit 11. The filter 13 outputs a first audio signal SV1' that does not include the low-frequency components. Therefore, the first audio signal SV1' is an audio signal that does not include a first formant component, but does include second, third, ..., nth order formant components.

[0042] The amplifier 14 amplifies the first voice signal SV1' that has passed through the filter 13, and outputs a first voice signal SV1'' in which the intensities of the second formant components, third formant components, ..., nth formant components have been increased.

[0043] The adjuster 15 adjusts the gain of the amplifier 14. In this embodiment, the adjuster 15 is configured using a switch capable of adjusting the gain to three levels from 0 to 2. The adjuster 15 is configured so that when "0" is selected by the switch, the gain becomes 1x, when "1" is selected by the switch, the gain becomes 3x, and when "2" is selected by the switch, the gain becomes 5x.

[0044] However, the number of stages of the switches constituting the adjuster 15 and the gain selected by the switches are not limited to the above example, and can be appropriately selected. For example, the gain may be configured such that when "0" is selected by the switch, the gain is 0 times, when "1" is selected by the switch, the gain is 1 time, and when "2" is selected by the switch, the gain is 5 times. When the gain is set to 0 times, the voice processing in the voice processor 10 is substantially canceled. Therefore, the voice processing device 1 converts the voice signal supplied from the microphone 21 into voice and outputs it from the speaker without performing voice processing for clarifying the voice. When it is known that voice clarifying processing is not necessary, the voice processing may be canceled as described above. Also, normally, the voice processing is used with the voice processing canceled, and when a gesture that seems difficult for the listener to hear is recognized, the speaker can adjust the adjuster 15 appropriately to adjust the degree of voice processing in the voice processor 10. Also, the adjuster 15 can adopt a volume that changes the gain continuously instead of a switch that changes the gain discretely. Furthermore, when the users who will be listeners can be identified to a certain extent, the gain is preset and adjuster 15 can be omitted.

[0045] The phase corrector 12 corrects the phase of the second audio signal SV2 so that it matches the phase of the first audio signal SV1" that has passed through the amplifier 14. In other words, the phase corrector 12 outputs the second audio signal SV2' that is in phase with the first audio signal SV1".

[0046] The additive synthesizer 16 generates a synthetic audio signal SSV by additively synthesizing the first audio signal SV1″ passed through the amplifier 14 and the second audio signal SV2′ passed through the phase corrector 12, and supplies the synthetic audio signal SSV to a port P12.

[0047] The port P12 is a port connected to a speaker and is an example of a second port. The port P12 and the speaker are connected using a cable or wiring. The composite voice signal SSV is supplied from the additive synthesizer 16 to the terminal of the port P12.

[0048] In this manner, the voice processor 10 performs voice processing on the voice signal SV supplied from the microphone 21 and supplies the synthesized voice signal SSV to the speaker 22 .

[0049] In this embodiment, the microphone 21 supplies the voice signal SV, which is an analog signal, to the port P11. The port P12 supplies the synthesized voice signal SSV, which is an analog signal, to the speaker. However, the microphone 21 may be configured to supply the voice signal SV, which is a digital signal, to the port P11, and the port P12 may be configured to supply the synthesized voice signal SSV, which is a digital signal, to the speaker.

[0050] In this embodiment, each of the phase corrector 12, the filter 13, the amplifier 14, and the additive combiner 16 is configured with an analog circuit. However, each of the phase corrector 12, the filter 13, the amplifier 14, and the additive combiner 16 may be configured with a digital circuit.

[0051] In the present embodiment, the audio processor 10 includes a filter 13 that is a high-pass filter for filtering the first audio signal SV1. However, the audio processor 10 may be configured such that the filter 13 removes (1) low-frequency components having a frequency of less than 400 Hz and (2) high-frequency components having a frequency of more than 7 kHz from the first audio signal SV1 that has passed through the branching unit 11. That is, the filter 13 may be a band-pass filter that passes components having a frequency of 400 Hz or more and 7 kHz or less, or may be a band-pass filter that passes components having a frequency of 400 Hz or more and 5 kHz or less. In this case, the filter 13 may be realized as one band-pass filter, or may be realized by connecting a high-pass filter and a low-pass filter in series.

[0052] In addition, in this embodiment, the housing of the audio processor 10 or the audio processing device 1 is made of aluminum. However, the material constituting the housing is not limited to aluminum, and may be, for example, a metal such as copper or stainless steel. The material constituting the housing may be mainly made of resin with a metal layer provided on the surface. Since at least the surface of the housing is covered with metal, it is possible for the housing to block electromagnetic waves that invade from the outside to the inside.

[0053] In one aspect of the present invention, a single housing that houses the sound processor 10 may be configured to house at least one of the microphone 21 and the speaker 22 in addition to the sound processor 10. In this embodiment, as shown in FIG. 2, the housing is configured to house the sound processor 10 and the speaker 22. Note that the sound processor 10, the microphone 21, and the speaker 22 are housed in a single housing, and the sound processing device 1 that is portable can be said to be a type of device called an electronic megaphone or loudspeaker. In addition, when it is assumed that a high-volume sound is output and a high-output speaker is used as the speaker 22, the sound processor 10 and the speaker 22 may be housed in a single housing, and the microphone 21 may be an external device. However, even in such a case, the sound processing device 1 including the sound processor 10, the speaker 22, and the housing is limited in weight and size to be portable.

[0054] The power supply unit, the detection unit, and the control unit, which are not shown in FIG. 1, are each configured as follows.

[0055] The power supply unit supplies power to the voice processing unit (particularly the amplifier 14) and the control unit. In order to configure the voice processing device 1 to be portable, it is preferable that the power supply unit is a battery. By using a battery as the power supply unit, the speaker can speak while moving freely without worrying about the power cable. However, assuming that the speaker's range of movement is limited, an AC / DC converter, such as an AC adapter, may be used as the power supply unit. By using an AC / DC converter as the power supply unit, a commercial power source can be used, so that the speaker can speak without worrying about the remaining power. The power supply unit may also employ both a battery and an AC / DC converter.

[0056] The detection unit is configured to detect whether or not the power is being supplied from the power supply unit to the audio processing unit. If the power supply unit is functioning normally, the detection unit supplies detection information indicating "Yes" to the control unit. On the other hand, if the power supply unit is not functioning normally, the detection unit supplies detection information indicating "No" to the control unit. If the power supply unit is an AC / DC converter, an example of a state in which the power supply unit is not functioning normally is a power outage. If the power supply unit is a battery, an example of a state in which the power supply unit is not functioning normally is a dead battery.

[0057] The control unit controls the selection unit to apply or not apply audio processing to the audio signal SV according to the detection result of the detection unit. When the detection result of the detection unit is "Yes", the control unit controls the selection unit to apply audio processing to the audio signal SV. On the other hand, when the detection result of the detection unit is "No", the control unit controls the selection unit to not apply audio processing to the audio signal SV.

[0058] Moreover, the audio processor 10 may further include a polarity switch and an amplifier interposed between the port P11 and the branching section 11.

[0059] (speaker) The speaker 22 converts the synthesized voice signal SSV supplied from the port P12 into voice and outputs the voice. In this embodiment, the voice signal supplied from the port P12 is an analog signal and an electrical signal. However, the voice signal supplied from the port P12 may be a digital signal, and the speaker 22 may be configured to convert the digital voice signal into voice. Alternatively, the speaker 22 may have a function of amplifying the voice, and the voice processor 10 may be realized as a loudspeaker.

[0060] (External view of the audio processing device) The left diagram in FIG. 2 is an example of an external view of the voice processing device 1, and the right diagram is a schematic diagram of a user wearing the voice processing device 1. As shown in FIG. 2, for example, the microphone 21 and the voice processor 10 may be connected by wire or wirelessly. In the example of FIG. 2, the microphone 21 is connected to a support member 24 for positioning the microphone 21 at the mouth of the user. The support member 24 is worn along the upper part of the ear and the back of the head of the user. In addition, the voice processor 10 may be fixed to a belt or the like and may be configured to be integrated with the microphone 21. According to the configuration of FIG. 2, the voice processing device 1 can be used in a hands-free manner, improving the convenience of the user.

[0061] <Frequency characteristics of audio processing device> FIG. 3 is a graph showing frequency characteristics in the audio processing device 1 equipped with the audio processor 10. The characteristic line a shown in FIG. 3 shows the frequency characteristics of the audio signal SV supplied from the microphone 21. The characteristic line b shown in FIG. 3 shows the frequency characteristics of the audio signal SV after its output value has been adjusted by the amplifier interposed between the port P11 and the branching section 11. The characteristic line c shown in FIG. 3 shows the frequency characteristics of the first audio signal SV1″ after passing through the amplifier 14. The characteristic line d shown in FIG. 3 shows the frequency characteristics of the synthesized audio signal SSV. The difference between the characteristic lines b and d, indicated by the arrows in FIG. 3, is the effect of the audio processing performed by the audio processor 10.

[0062] The output level of the frequency characteristic of the first audio signal SV1'' represented by the characteristic line c can be adjusted by adjusting the gain of the amplifier 14 using the adjuster 15.

[0063] <Effects of audio processing device> The voice processing device 1 is a conversation assistance device equipped with a function that provides a method for reliably conveying the contents of a conversation in face-to-face conversations with people with hearing impairments. This method was born from an idea completely opposite to that of conventional hearing aids and sound collectors. The speaker emits a voice that has been processed to make it easier for the hearing impaired to hear, via the voice processing device 1, making it possible for the hearing impaired to convey the contents of the conversation without using a hearing aid or the like.

[0064] The voice processing technology for voice clarity installed in the voice processing device 1 focuses on the fact that the efficiency of the cochlea, the auditory organ of the inner ear, which causes hearing impairment due to aging, in converting voice into electrical signals, especially in the high frequency band (called formant), is reduced, hindering the understanding of language, and adopts the feature of being able to restore hearing by performing processing that emphasizes the high frequency components contained in words. In the spectrum analysis of voice, the peak frequency with the highest level that appears at the lowest frequency is called the first formant, the second peak frequency is called the second formant, etc., and each peak frequency appears at an integer multiple point, and the peak frequency differs depending on the speaker's bone structure, etc., but it has been clarified by our many findings over many years that the formant components necessary for understanding language generally need to be reliably heard from the first formant to the fourth formant, and it has been confirmed that the voice processing device 1 can demonstrate its effect by focusing on compensation of 400Hz to 5KHz.

[0065] As described above, the audio processor 10 of the audio processing device 1 includes the port P11, the branching unit 11, the filter 13, the amplifier 14, the phase corrector 12, the additive combiner 16, and the port P12.

[0066] According to the above configuration, the amplifier 14 amplifies the first voice signal SV1' and does not amplify the second voice signal SV2. Therefore, the voice processor 10 can output a synthetic voice signal in which the first formant components are not amplified and predetermined formant components (e.g., second, third, and fourth formant components) are amplified. Therefore, the voice processing device 1 can output a synthetic voice signal SSV that the listener feels is clear without using an auxiliary device.

[0067] As a result, the possibility of misunderstandings occurring between the speaker and the listener or of upsetting the listener can be reduced, making it easier for communication to proceed smoothly between the speaker and the listener.

[0068] The inventor has found that, when clarifying a voice, it is preferable to remove (1) low-frequency components having a frequency of less than 400 Hz and (2) high-frequency components having a frequency of more than 7 kHz from the first voice signal SV1, amplify the first voice signal SV1', and add and synthesize the first voice signal SV1" and the second voice signal SV2, which are in phase with each other. However, the wavelength of a human voice is usually about 100 Hz or more and 1 kHz or less. Therefore, in the voice processor 10 constituting the voice processing device 1, it is often sufficient for the filter 13 to be able to remove only the low-frequency components from the first voice signal SV1. In other words, the filter 13 does not need to be configured to remove the high-frequency components from the first voice signal SV1. Therefore, the filter 13 can be configured simply, and costs can be reduced.

[0069] In the audio processing device 1, each of the phase corrector 12, the filter 13, the amplifier 14, and the additive synthesizer 16 is preferably configured by an analog circuit.

[0070] According to the above configuration, the voice processor 10 can be easily configured and can be made smaller. Furthermore, compared to a case where the filter 13 is configured with a digital circuit, the voice processing device 1 configured in this manner can output a synthetic voice signal SSV that the listener feels is clearer. This is considered to be because the filter 13 configured with an analog circuit is more likely to have gentler filtering characteristics in the lower limit region of the passband (i.e., the region around 400 Hz) than the filter 13 configured with a digital circuit.

[0071] In the audio processing device 1, the audio processor 10 preferably further includes an adjuster 15.

[0072] According to the above configuration, the speaker user can adjust the gain of the amplifier 14 according to the state of the listener user, regardless of the listener user's intention. That is, the degree of clarity in the synthetic speech signal can be adjusted. Therefore, the speech processing device 1 configured in this way can output a synthetic speech signal SSV with the degree of clarity adjusted according to the listener user.

[0073] In the voice processing device 1, the filter 13 is preferably configured to remove the low-frequency components below a lower limit frequency and the high-frequency components above an upper limit frequency. The band between the lower limit frequency and the upper limit frequency may include second, third, and fourth formant components.

[0074] The inventors of the present application have found that the frequency band that contributes to the clarity of an audio signal is approximately between 400 Hz and 7 kHz. According to the audio processing device 1 configured in this manner, the amplifier 14 amplifies the first audio signal SV1' that is included in the frequency band that contributes to the clarity of an audio signal.

[0075] The audio processor 10 included in the audio processing device 1 is also one aspect of the present invention.

[0076] The voice processor 10 can output a synthetic voice signal in which the first formant components are not amplified and predetermined formant components (eg, second, third and fourth formant components) are amplified.

[0077] Furthermore, in the audio processor 10, each of the audio signal SV and the synthesized audio signal SSV is preferably an analog signal.

[0078] In addition, in the audio processor 10, each of the phase corrector 12, the filter 13, the amplifier 14, and the additive synthesizer 16 is preferably configured by an analog circuit.

[0079] This allows the voice processor 10 to be easily configured and to be made smaller in size. Furthermore, compared to a case in which the filter 13 is configured using a digital circuit, the voice processor 10 configured in this manner can output a synthetic voice signal that the listener perceives as clearer.

[0080] [Embodiment 2] Other embodiments of the present invention will be described below. For ease of explanation, the same reference numerals are given to members having the same functions as those described in the above embodiment, and the description thereof will not be repeated.

[0081] <Configuration of voice processing device> As shown in FIG. 4, the voice processing device 1A includes a microphone 21, a speaker 22, and a housing similar to those in the first embodiment, as well as a voice processor 10A.

[0082] [Configuration of audio processing device] The audio processor 10A includes an audio signal generator 11A, a correction signal generator 12A, a filter 13A, and a synthesis unit 14A. The audio processor 10A according to the present embodiment further includes a first port 15A, an input buffer unit 16A, an adjustment unit 17, and a second port 18. The audio processor 10A may further include a power supply unit, a detection unit, and a control unit, which are not shown.

[0083] (1st port) The first port 15A outputs an audio signal x representing a voice supplied from a device connected to the first port 15A to the input buffer unit 16A. As shown in FIG. 5, the audio signal x includes a first formant component h1, a second formant component h2, ..., and an nth formant component hn. Note that n depends on the user on the transmitting side and is a positive integer of at least 4, although there may be some individual differences. Note that the first formant component h1, the second formant component h2, ..., and the nth formant component hn are described in Patent Document 1, and therefore will not be described in this embodiment.

[0084] (Input buffer) As shown in FIG. 4, the input buffer unit 16A is provided between the first port 15A and the audio signal generating unit 11A. The input buffer unit 16A can be configured with a microphone amplifier, a preamplifier for lowering output impedance, etc., depending on the usage situation. The input buffer unit 16A according to this embodiment adjusts the audio signal x to narrow the upper limit and the adjustment of the frequency and to increase the sound pressure, as shown in FIG. 6, for example. Then, the input buffer unit 16A outputs the audio signal x that has been subjected to the necessary adjustment to the audio signal generating unit 11A, as shown in FIG. 4. Note that, when the adjustment of the audio signal supplied from the first port is not required (the device connected to the first port is configured to output an audio signal that does not require adjustment), the audio processor 10A does not need to include the input buffer unit 16A.

[0085] (Audio signal generation section) The audio signal generating unit 11A generates a first audio signal x1 and a second audio signal x2 having the same spectrum from the audio signal x supplied from the input buffer unit 16A. The "having the same spectrum" means that the first audio signal x1 and the second audio signal x2 each contain a first-order formant component h1, a second-order formant component h2, ..., and an n-order formant component hn. The audio signal generating unit 11A according to this embodiment further generates a third audio signal x3 having the same spectrum as the second audio signal x2. The audio signal generating unit 11A then supplies the first audio signal x1 to the high-pass filter 131, the second audio signal x2 to the synthesis unit 14A, and the third audio signal to the correction signal generating unit 12A. The audio signal generating unit 11A according to this embodiment generates the first audio signal x1, the second audio signal x2, and the third audio signal x3 by branching the audio signal. In this embodiment, the audio signal generating unit 11A is configured so that the intensity ratio between the first audio signal x1, the second audio signal x2, and the third audio signal x3 is 1:1:1, that is, the distribution ratio is 1:1:1. However, the distribution ratio of the audio signals is not limited to 1:1:1 and can be set appropriately. The audio signal generating unit 11A may also copy the audio signal supplied from the input buffer unit 16A and set the original audio signal as the first audio signal x1, and the copied audio signal as the second audio signal x2 and the third audio signal x3.

[0086] (Correction signal generation unit) The correction signal generating unit 12A generates a phase correction signal p based on the second audio signal x2 supplied from the audio signal generating unit 11A. As described above, the audio signal generating unit 11A according to this embodiment generates a third audio signal x3 having the same phase as the second audio signal x2. Therefore, the correction signal generating unit 12A according to this embodiment generates a phase correction signal p based on the third audio signal x3. As a result, the correction signal generating unit 12A generates a phase correction signal p based on a dedicated third audio signal. Therefore, it is possible to prevent the second audio signal from being out of phase by using the second audio signal to generate the phase correction signal p. Then, the correction signal generating unit 12A outputs the generated phase correction signal p to the operational amplifier 132a of the filter 13A.

[0087] As shown in FIG. 7, the correction signal generating unit 12A according to the present embodiment includes a capacitor 121 and a resistor 122. One end of the capacitor 121 is connected to the audio signal generating unit 11A, and the other end is connected to the non-inverting input terminal (+) of the operational amplifier 132a. One end of the resistor 122 is connected to the other end of the capacitor 121, and a bias voltage is applied to the other end. The correction signal generating unit 12A thus configured outputs a phase correction signal p having only frequency components equal to or higher than 1 / 2πCR (C: electrostatic capacitance, R: resistance value) by a high-pass filter function internally configured by the capacitor 121 and the resistor 122. Hereinafter, the lower limit value of the frequency components of the phase correction signal p is referred to as a phase correction start frequency fs. The correction signal generating unit 12A can adjust the phase correction start frequency fs by changing at least one of the capacitor 121 and the resistor 122. As will be described later in detail, the degree to which the operational amplifier 132a of the filter 13 corrects the phase varies within a range of 0 to -180° depending on the phase correction start frequency fs of the phase correction signal p input to the non-inverting input terminal (+) as shown in Fig. 5. For this reason, the phase correction start frequency fs is adjusted to an optimum value that minimizes the phase difference between the corrected first audio signal x1 and second audio signal x2.

[0088] Furthermore, the impedance Z (internal resistance) of the capacitor 121 changes depending on the phase correction start frequency fs. Specifically, the impedance Z of the capacitor 121 is inversely proportional to the phase correction start frequency fs, and its value is expressed as 1 / ωC ​​(ω=2πfs). That is, the impedance Z of the capacitor 121 decreases as the phase correction start frequency fs increases. Therefore, the correction signal generating unit 12A outputs a phase correction signal p with a higher voltage as the phase correction start frequency fs increases, and outputs a phase correction signal p with a lower voltage as the phase correction start frequency fs decreases. In this way, by using the capacitor 121, the correction signal generating unit 12A can be configured to be compatible with audio signals of various frequencies with a simple configuration.

[0089] (filter) The filter 13A removes at least a first frequency component equal to or lower than a predetermined first frequency from the first audio signal. The first frequency is a frequency between an upper limit frequency of a first formant component h1 and a lower limit frequency of a second formant component h2. The filter 13A according to this embodiment is a band-pass filter that also removes a second frequency component equal to or higher than a second frequency higher than the first frequency. The filter 13A according to this embodiment is composed of a high-pass filter 131 and a low-pass filter 132.

[0090] The high-pass filter 131 removes a first frequency component from the first audio signal x1 supplied from the audio signal generating unit 11A. In the high-pass filter 131 according to the present embodiment, the first frequency is set to 400 Hz. The high-pass filter 131 outputs the first audio signal x1' from which the first frequency component has been removed, that is, an audio signal that does not include the first formant component h1 and includes the second formant component h2, the third formant component h3, . . . , the nth formant component hn, to the low-pass filter 132.

[0091] The high-pass filter 131 according to this embodiment has an operational amplifier (not shown) that amplifies the first audio signal x1. Therefore, the high-pass filter 131 according to this embodiment outputs the first audio signal x1 from which the first frequency component has been removed and which has been amplified. The operational amplifier amplifies the first audio signal x1 from which the first frequency component has been removed, thereby outputting the first audio signal x1" in which the intensities of the second-order formant component h2, the third-order formant component h3, ..., and the n-th order formant component hn have been increased. Note that the high-pass filter 131 may be configured so that at least one of the gain of the operational amplifier and the frequency that is the boundary between removal and non-removal can be set arbitrarily depending on the purpose of use.

[0092] The low-pass filter 132 removes a second frequency component from the first audio signal x1, from which the first frequency component has been removed and which is supplied from the high-pass filter 131. The second frequency component is a frequency component equal to or higher than the second frequency. The second frequency (cutoff frequency) is a frequency between the upper limit frequency of the fifth formant component h5 and the lower limit frequency of the sixth formant component h6, or a frequency between the upper limit frequency of the sixth formant component h6 and the lower limit frequency of the seventh formant component h7. In the low-pass filter 132 according to this embodiment, the second frequency is set to 5 kHz or 7 kHz. Low-pass filter 132 then outputs first speech signal x1" from which the fifth, sixth and higher frequency components have been removed, i.e., a speech signal including only second- to fifth-order formant components h2-h5 or only second- to sixth-order formant components h2-h6. Higher-order formant components become noise that does not contribute to language understanding. For this reason, by low-pass filter 132 removing the second frequency component, it is possible to improve the signal-to-noise ratio in synthetic speech signal x', which will be described later.

[0093] However, when removing frequency components, a phase delay inevitably occurs in the first voice signal. If the first voice signal x1, whose phase is delayed by time T, is synthesized with the second voice signal x2 without correction, the voltage V1 of the first voice signal x1 becomes -V1 due to the phase delay, as shown in FIG. 9. Therefore, the voltage V of the synthesized voice signal becomes V2+(-V1) because the first voice signal x1 and the second voice signal x2 cancel each other out. As a result, as shown in FIG. 10, a dead zone (deep valley) called a dip occurs in a certain frequency range, and the sound pressure of the first formant component h1 and the second formant component h2, which are particularly important for language understanding, becomes extremely small, and other inconveniences occur that are a major obstacle to improving clarity.

[0094] For this reason, the filter 13A has a function of correcting the phase delay. Specifically, as shown in FIG. 7, an operational amplifier 132a is provided in the low-pass filter 132 together with other elements 132b and 132c. The operational amplifier 132a amplifies the first audio signal x1. The operational amplifier 132a is configured so that the first audio signal is input to its inverting input terminal (-). The operational amplifier 132a outputs a first audio signal x1' whose phase is corrected to approach the phase of the second audio signal x2 by inputting a phase correction signal p to its non-inverting input terminal (+). Specifically, the operational amplifier 132a compares and calculates both the voltage of the first audio signal input to its inverting input terminal (-) and the voltage of the phase correction signal p input to its non-inverting input terminal (+). If the voltage of the phase correction signal p is higher, the operational amplifier 132a outputs the first audio signal x1' whose phase is advanced according to the voltage difference. On the other hand, if the voltage of the phase correction signal p is lower, the first audio signal x1' whose phase is delayed according to the voltage difference is output. As described above, the voltage of the phase correction signal p changes according to the phase correction start frequency fs determined by the capacitor 121 and the resistor 122 of the correction signal generating unit 12A. Therefore, the degree to which the operational amplifier 132a advances or delays the phase varies within a range of 0 to -180° depending on the phase correction start frequency fs, as shown in FIG. 8, for example. As described above, the phase correction start frequency fs is adjusted in advance. Therefore, the phase correction signal p whose voltage corresponds to the adjusted phase correction start frequency fs is input to the non-inverting input terminal (+) of the operational amplifier 132a. Then, the operational amplifier 132a outputs the first audio signal x1' corrected so that the difference in phase with the second audio signal x2 is minimized.

[0095] The filter 13A may be configured such that the low-pass filter 132 first removes the second frequency component from the first audio signal, and the high-pass filter 131 removes the first frequency component from the first audio signal x1 from which the second frequency component has been removed. In this case, the phase correction signal p is input to the non-inverting input terminal (+) of the operational amplifier included in the high-pass filter 131.

[0096] (adjustment section) The adjustment unit 17 adjusts the sound pressure of the first audio signal x1' whose phase has been corrected. Then, the adjustment unit 17 supplies the adjusted first audio signal x1' to the synthesis unit 14A. The adjustment unit 17 adjusts the first audio signal based on a user's setting operation according to the usage environment and the degree of hearing loss. Therefore, the adjustment unit 17 can adjust only the first to fifth formant components h1 to h5, which are important for language understanding, to a level exceeding the masking line MK', which is an audible level. In addition, the adjustment unit 17 can freely select the output level curve of the synthetic audio signal x' to be "minimum", "maximum", or "middle" as shown in FIG. 11.

[0097] (Synthetic section) The synthesis unit 14A generates a synthetic audio signal x' by additively synthesizing the first audio signal x1', from which the first frequency component and the second frequency component have been removed and the phase has been corrected, and the second audio signal x2, supplied from the audio signal generation unit 11A. As described above, the audio processor 10A according to the present embodiment includes an adjustment unit 17. For this reason, the synthesis unit 14A according to the present embodiment additively synthesizes the first audio signal x1, from which the sound pressure has been adjusted, and the second audio signal x2. Then, the synthesis unit 14A supplies the generated synthetic audio signal x' to the second port 18. As shown in FIG. 12, the phase of the corrected first audio signal x1' in the synthesis unit 14A leads the phase of the first audio signal x1 before correction by T', so that the corrected first audio signal x1' is in phase with the second audio signal x2. As a result, the frequency characteristic of the synthetic audio signal x' output by the synthesis unit 14A becomes as shown in FIG. 13.

[0098] (2nd port) The second port 18 outputs the synthesized voice signal x′ supplied from the synthesis unit 14A to a device connected to the second port 18.

[0099] (Power supply part) The power supply unit supplies power to each unit requiring power in the audio processor 10A. The type of the power supply unit is not limited, and may be an AC / DC converter such as an AC adapter, or a battery.

[0100] (Detection unit) The detection unit is configured to detect whether or not power is being supplied from the power supply unit to the audio processing unit. If the power supply unit is functioning normally, the detection unit supplies detection information indicating "Yes" to the control unit. On the other hand, if the power supply unit is not functioning normally, the detection unit supplies detection information indicating "No" to the control unit. If the power supply unit is an AC / DC converter, an example of a state in which the power supply unit is not functioning normally is a power outage. If the power supply unit is a battery, an example of a state in which the power supply unit is not functioning normally is a dead battery.

[0101] (Selection section) The selection unit selects whether or not to apply audio processing to the audio signal x based on the control of the control unit. The selection unit is provided, for example, between the low-pass filter 132 and the synthesis unit 14A, and is configured by a switch that switches the circuit between ON / OFF. When the selection unit is ON, the first audio signal x1 that has passed through the high-pass filter 131 and the low-pass filter 132 is supplied to the synthesis unit 14A, and is synthesized with the second audio signal x2 in the synthesis unit 14A. On the other hand, when the selection unit is OFF, the first audio signal x1 is not supplied to the synthesis unit 14A, and the second audio signal x2 is output from the synthesis unit 14A. Note that the selection unit may be provided between the audio signal generation unit 11A and the high-pass filter 131, or between the high-pass filter 131 and the low-pass filter 132.

[0102] (Control unit) The control unit controls the selection unit to apply or not apply audio processing to the audio signal x according to the detection result of the detection unit. When the detection result of the detection unit is "Yes", the control unit controls the selection unit to apply audio processing to the audio signal x (turn ON). On the other hand, when the detection result of the detection unit is "No", the control unit controls the selection unit to not apply audio processing to the audio signal x (turn OFF).

[0103] [Variations of the voice processing device] In the above embodiment, the high-pass filter 131, the low-pass filter 132, the correction signal generating unit 12A, and the synthesis unit 14A of the audio processor 10A are configured as analog circuits. However, at least any of these may be configured as digital circuits.

[0104] Moreover, the audio processor 10A according to the above embodiment includes the high-pass filter 131 and the low-pass filter 132. However, instead of including these filters, the audio processor 10A may include a band-pass filter that simultaneously removes the first frequency component and the second frequency component.

[0105] Moreover, the audio processor 10A may further include a polarity switch and an amplifier interposed between the first port 15A and the audio signal generator 11A.

[0106] The audio processor 10A also includes a selection unit, an amplified audio signal processor, and a selector unit, which are disposed between the synthesis unit 14A and the second port 18. The amplifier may further include a circuit isolator, a transmitter power regulator, and a polarity switch.

[0107] Furthermore, the audio processor 10A may include an attenuator between the synthesis unit 14A and the second port 18. The attenuator arbitrarily adjusts the output in accordance with the input audio source signal. By including such an attenuator, it becomes possible to provide a system that can accommodate hearing loss symptoms due to hearing impairment by incorporating the microphone line of an existing wired or wireless audio output device, or the signal line of a television, radio, or other entertainment playback device.

[0108] [Effects of the audio processing device] The above-described audio processor 10A corrects the phase delay that occurs when the low-pass filter 132 removes the first and second frequency components based on the phase correction signal p. Therefore, the audio processor 10A can make the audio even clearer than before.

[0109] [Modifications] The voice processors 10 and 10A may be realized, for example, so as to include the configurations described in the patent publications related to the following patent numbers, or in combination with the configurations. The technologies described in each of the patent publications of Japanese Patent No. 3731179 and Japanese Patent No. 6548938 are technologies for implementing voice processing for voice clarity by focusing on first-order (or low-order) formant components and high-order formant components (see, for example, FIG. 1 of Japanese Patent No. 3731179 and FIG. 4 of Japanese Patent No. 6548938). Therefore, instead of the configuration shown in the block diagram of FIG. 5, the configurations shown in FIG. 2, FIG. 4, and FIG. 6 of Japanese Patent No. 3731179 may be adopted as the configuration for realizing the voice processor 10, or the configuration shown in FIG. 5 of Japanese Patent No. 6548938 may be adopted. The audio processor 10 is required only to perform audio processing on an audio signal representing the voice of the speaking party supplied from the microphone 21 in order to clarify the audio, and to supply the processed audio signal to the speaker 22.

[0110] Moreover, instead of the audio processor 10 shown in FIG. 5, the configuration shown in FIG. 4 of Japanese Patent No. 7105756 can be used.

[0111] [Additional Notes] The present invention is not limited to the above-described embodiments, and various modifications are possible within the scope of the claims. Embodiments obtained by appropriately combining the technical means disclosed in the description of the invention are also included in the technical scope of the present invention. [Explanation of symbols]

[0112] 1 Audio processing device 10 Audio processor 11 Branch 12 Phase corrector 13 Filter (Bandpass Filter) 14 Amplifier 15 Regulator 16 Additive Synthesizer P11 port (first port) P12 port (second port) 21. Mike 22 Speakers 1A Audio Processing Device 10A Audio processor 10, 10A Audio processor 11A Audio signal generator 12A correction signal generation section 121 Capacitor 122 resistor 13A Filter 131 High Pass Filter 132 Low-pass filter 132a Operational Amplifier 132b Element 14A Synthesis section 15A Port 1 16A Input Buffer 17 Adjustment part 18 Second Port 21. Mike 22 Speakers 24 Support member

Claims

1. 1. A portable audio processing device, comprising: an audio processor that performs audio processing on the audio signal supplied from the microphone to make the audio clear; a speaker that converts the processed audio signal into audio; and a housing that houses at least the audio processor.

13. A voice processing device comprising:

2. The housing houses the speaker in addition to the audio processor.

2. The audio processing device according to claim 1 .

3. Further comprising a microphone for converting voice into an audio signal; The housing accommodates the microphone in addition to the audio processor and the speaker.

3. The audio processing device according to claim 2.

4. Further comprising a microphone for converting voice into an audio signal.

3. The audio processing device according to claim 1, wherein the audio processing device is a digital audio processing device.

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