Sound processing device

Abstract

The present invention provides a sound processing device (1), which is portable and according to which a speaker uses an auxiliary device and so that a hearing-impaired person can easily listen to sound without using an auxiliary device. The sound processing device (1) comprises: a sound processor (10) for performing sound processing to make sound clear for a sound signal supplied from a microphone (21); a speaker (22) that converts the sound signal having been subjected to the sound processing into sound; and a housing that houses the sound processor (10).

Description

Audio Processing Device

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

[0002] BACKGROUND ART The proportion of the population suffering from hearing loss has been increasing due to the aging population, and assistive devices such as hearing aids and sound amplifiers are now commonly used.

[0003] Japanese Patent Application Publication No. 2016-110050

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

[0005] However, the voice processing device described in Patent Document 1 is primarily targeted at audio equipment and broadcasting facilities installed in places where large numbers of 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 pre-installed in places where large numbers of people gather. Therefore, this voice processing device is not portable, and therefore cannot be used in a state where the speaker and the voice processing device are mobile 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.

[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 that 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 houses at least the audio processor.

[0008] 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.

[0009] 1 is a block diagram of an audio processing device according to a first embodiment of the present invention. The left diagram is an example of an external view of the audio processing device shown in FIG. 1, and the right diagram is a schematic diagram of a user wearing the audio processing device shown in FIG. 1. A graph showing frequency characteristics in the audio processing device shown in FIG. 1. A block diagram showing an example of the configuration of an audio processing device according to a second embodiment of the present invention. A diagram explaining formant components contained in an audio signal. A graph showing adjustment characteristics of an input buffer unit included in the audio processing device of FIG. 4. A circuit diagram showing an example of a specific configuration of a filter and a correction signal generation unit included in the audio processing device of FIG. 4. A graph showing phase correction characteristics of a filter included in the audio processing device of FIG. 4. A graph explaining a case where a first audio signal and a second audio signal are synthesized without phase correction. A graph showing frequency characteristics of a synthesized audio signal when phase correction is not performed. A graph showing an example of adjustment of an audio signal by an adjustment unit included in the audio processing device of FIG. 4. A graph showing changes over time in phase and output of an audio signal, a first audio signal before and after correction, and a synthesized audio signal. A graph showing overall output characteristics of a synthesis unit.

[0010] [First Embodiment] Hereinafter, one embodiment of the present invention will be described in detail.

[0011] <Outline of speech processing device> In recent years, with the aging of the population, it has been reported that there are frequent cases in which important information spoken by a speaker is not conveyed to a listener with hearing loss. The general idea is to simply increase the speaker's volume. However, there are two types of hearing loss: conductive hearing loss and sensorineural hearing loss caused by damage to the inner ear. Sensorineural hearing loss in particular makes it impossible to understand speech even at high volumes. It has been discovered that the cause of this is a decline in the ability of the cochlea, the organ of hearing, to convert high frequencies into auditory signals.

[0012] When hearing function declines due to aging or other factors, conversational communication becomes less common in everyday life. For this reason, hearing aids and sound amplifiers are commonly used by people with hearing loss. These devices boost the volume and apply some frequency processing to the sound picked up by the microphone, making it easier to hear, and output the sound from earphones.

[0013] 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 have all of the hearing-impaired people wear an assistive device.

[0014] Therefore, the inventors have sought a method for enabling a speaker to use an assistive device to make it easier for a hearing-impaired person to hear speech without using an assistive device.

[0015] 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 the frequency characteristics of the voice processing device 1.

[0016] <Idea of ​​the speech processing device> The speech processing device 1, which is a type of assistive device, is equipped with technology that supplementarily corrects frequency components that play an important role in understanding language. This enables the speech processing device 1 to accurately understand language that was previously difficult to hear. Furthermore, the speech processing device 1 is realized using a method that is completely opposite to the conventional idea of ​​correcting speech signals on the speaker's side.

[0017] 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. This function makes it possible to realize a smooth conversation.

[0018] 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 accommodates at least the audio processor 10.

[0019] 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 speaker's body or clothing, or may be configured to have a holder that can be held by the speaker.

[0020] In this embodiment, the voice processing device 1 includes a clip 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 pants, a skirt, or a belt worn by the speaker. Therefore, the voice processing device 1 is configured so that the speaker can carry it while wearing it around their 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 clip described above, 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.

[0021] 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. The grip part of the electronic megaphone that is held by a speaker is an example of a holding part that can be held by a speaker.

[0022] 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. Furthermore, if 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.

[0023] (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.

[0024] For example, an electric condenser microphone can be used as the microphone 21. A DC voltage is applied to the electric condenser microphone for driving it. Furthermore, electric condenser microphones are often connected using connectors of different types depending on the manufacturer or model.

[0025] (Audio Processor) An audio processor according to one aspect 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 undergone the audio processing. Below, an audio processor 10, which is an example of an audio processor according to one aspect of the present invention, will be described.

[0026] 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 disposed 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 also includes a power supply unit, a detection unit, and a control unit, which are not shown in Fig. 1.

[0027] 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 the voice of a speaker by the microphone 21 is supplied from the microphone 21 to a terminal of the port P11.

[0028] As shown in FIG. 1 of Patent Document 1, the speech signal SV includes first, second, ..., nth formant components. Note that n depends on the user on the transmitting side and, although there may be some individual differences, is a positive integer of at least 4. Note that the first, second, ..., nth formant components can be read as the first, second, ..., nth formant components shown in FIG. 1 of Patent Document 1, respectively.

[0029] The first-order formant components, second-order formant components, ..., n-order formant components are described in Patent Document 1. Therefore, in this embodiment, a description thereof will be omitted.

[0030] 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, i.e., so that 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 as appropriate.

[0031] The spectrum of each of the first audio signal SV1 and the second audio signal SV2 is similar to the spectrum of the audio signal SV. That is, each of the first audio signal SV1 and the second audio signal SV2 includes a first-order formant component, a second-order formant component, ..., an n-th order formant component.

[0032] The filter 13 is a high-pass filter that removes low-frequency components, that is, 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-order formant component, but does include second-order formant components, third-order formant components, ..., n-th order formant components.

[0033] The amplifier 14 amplifies the first audio signal SV1' that has passed through the filter 13, and outputs a first audio signal SV1'' in which the intensities of the second-order formant components, third-order formant components, ..., n-order formant components have been increased.

[0034] The adjuster 15 adjusts the gain of the amplifier 14. In this embodiment, the adjuster 15 is configured using a switch that can adjust the gain in three stages 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.

[0035] However, the number of switch stages constituting the adjuster 15 and the gain selected by the switch are not limited to the above example and can be selected as appropriate. For example, the adjuster 15 may be configured so that the gain is 0 when "0" is selected by the switch, the gain is 1 when "1" is selected by the switch, and the gain is 5 when "2" is selected by the switch. When the gain is set to 0, the voice processing in the voice processor 10 is essentially 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 to clarify the voice. If it is known that voice clarity processing is unnecessary, the voice processing may be canceled as described above. Furthermore, the voice processing device 1 may normally be used with the voice processing canceled. Then, if the listener's gestures that seem difficult to hear are recognized, the speaker can adjust the adjuster 15 as appropriate to adjust the degree of voice processing in the voice processor 10. Furthermore, a volume that continuously changes the gain can be used instead of a switch that discretely changes the gain as the adjuster 15. Furthermore, if the listener can be identified to some extent, the gain can be set in advance and the adjuster 15 can be omitted.

[0036] 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 a second audio signal SV2' that is in phase with the first audio signal SV1".

[0037] The additive synthesizer 16 generates a synthesized audio signal SSV by additively synthesizing the first audio signal SV1" that has passed through the amplifier 14 and the second audio signal SV2' that has passed through the phase corrector 12. The additive synthesizer 16 then supplies the synthesized audio signal SSV to a port P12.

[0038] 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 terminal of the port P12 is supplied with the synthesized voice signal SSV from the additive synthesizer 16.

[0039] In this way, the audio processor 10 performs audio processing on the audio signal SV supplied from the microphone 21. Then, the audio processor 10 supplies the generated synthesized audio signal SSV to the speaker 22.

[0040] In this embodiment, the microphone 21 supplies the audio signal SV, which is an analog signal, to the port P11. The port P12 supplies the synthesized audio signal SSV, which is an analog signal, to the speaker. However, the microphone 21 may be configured to supply the audio signal SV, which is a digital signal, to the port P11. Alternatively, the port P12 may be configured to supply the synthesized audio signal SSV, which is a digital signal, to the speaker.

[0041] In this embodiment, the phase corrector 12, the filter 13, the amplifier 14, and the additive combiner 16 are each configured with an analog circuit. However, the phase corrector 12, the filter 13, the amplifier 14, and the additive combiner 16 may each be configured with a digital circuit.

[0042] Furthermore, in this embodiment, the audio processor 10 includes the filter 13, which is a high-pass filter that filters the first audio signal SV1. However, in the audio processor 10, the filter 13 may be configured to remove (1) low-frequency components, which are components with a frequency of less than 400 Hz, and (2) high-frequency components, which are components with 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 with a frequency of 400 Hz or more and 7 kHz or less, or may be a band-pass filter that passes components with a frequency of 400 Hz or more and 5 kHz or less. In this case, the filter 13 may be realized as a single band-pass filter, or may be realized by connecting a high-pass filter and a low-pass filter in series.

[0043] 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 metal such as copper or stainless steel. The material constituting the housing may also be primarily resin with a metal layer provided on the surface. By covering at least the surface of the housing with metal, electromagnetic waves entering from the outside into the inside can be blocked.

[0044] In one aspect of the present invention, a single housing that houses the audio processor 10 may be configured to house at least one of a microphone 21 and a speaker 22 in addition to the audio processor 10. As shown in FIG. 2 , the housing according to this embodiment is configured to house the audio processor 10 and the speaker 22. A portable audio processing device 1 that houses the audio processor 10, the microphone 21, and the speaker 22 in a single housing can be considered a type of device called an electronic megaphone or loudspeaker. Furthermore, when a high-output speaker is used as the speaker 22 to output loud audio, the audio 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 audio processing device 1 including the audio processor 10, the speaker 22, and the housing is limited in weight and size to be portable.

[0045] The power supply unit, the detection unit, and the control unit, which are not shown in FIG. 1, are each configured as follows.

[0046] The power supply unit supplies power to the audio processing unit (particularly the amplifier 14) and the control unit. In order to make the audio processing device 1 portable, the power supply unit is preferably 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, commercial power can be used, so the speaker can speak without worrying about the remaining power. Furthermore, the power supply unit may employ both a battery and an AC / DC converter.

[0047] 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.

[0048] The control unit controls the selection unit to perform or not perform audio processing on the audio signal SV according to the detection result of the detection unit. If the detection result of the detection unit is "Yes", the control unit controls the selection unit to perform audio processing on the audio signal SV. On the other hand, if the detection result of the detection unit is "No", the control unit controls the selection unit not to perform audio processing on the audio signal SV.

[0049] The audio processor 10 may further include a polarity switch and an amplifier interposed between the port P11 and the branching unit 11.

[0050] (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 to amplify the voice, and the voice processor 10 may be realized as a loudspeaker.

[0051] (External View of Audio Processing Device) The left diagram in FIG. 2 is an example of an external view of the audio processing device 1, and the right diagram is a schematic diagram of a user wearing the audio processing device 1. As shown in FIG. 2, for example, the microphone 21 and the audio 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 near the user's mouth. The support member 24 is worn along the top of the user's ear and the back of the head. In addition, the audio processor 10 may be fixed to a belt or the like and be configured as an integrated part of the microphone 21. According to the configuration of FIG. 2, the audio processing device 1 can be used hands-free, thereby improving user convenience.

[0052] <Frequency characteristics of audio processing device> Figure 3 is a graph showing the frequency characteristics of the audio processing device 1 equipped with the audio processor 10. Characteristic line a shown in Figure 3 shows the frequency characteristics of the audio signal SV supplied from the microphone 21. Characteristic line b shown in Figure 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. Characteristic line c shown in Figure 3 shows the frequency characteristics of the first audio signal SV1" after passing through the amplifier 14. Characteristic line d shown in Figure 3 shows the frequency characteristics of the synthesized audio signal SSV. The difference between characteristic line b and characteristic line d, indicated by arrows in Figure 3, is the effect of the audio processing performed by the audio processor 10.

[0053] 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.

[0054] <Effects of the Voice Processing Device> The voice processing device 1 is a conversation assistance device equipped with a function that provides a method for reliably communicating the content of a conversation with a hearing-impaired person in face-to-face conversations. The voice processing device 1 generates voice using a method that is born from the completely opposite idea of ​​conventional hearing aids and sound collectors, in which voice processing is performed on the speaker to make it easier for the hearing-impaired person to hear. By having the speaker speak through such a voice processing device 1, the hearing-impaired person can communicate the content of the conversation without using a hearing aid or the like.

[0055] The speech processing-based speech clarity technology implemented in the speech processing device 1 addresses the issue of aging-related declines in the efficiency of the cochlea, the inner ear's auditory organ, in converting speech into electrical signals, particularly in the high-frequency band (known as formants), which impedes speech comprehension. For this reason, the speech processing device 1 incorporates a feature that emphasizes the high-frequency components contained in speech. This feature can restore hearing. In speech spectral analysis, the lowest peak frequency is called the first formant, the second peak frequency is called the second formant, and so on. The peak frequencies of each formant appear at integral multiples, but these peak frequencies vary depending on the speaker's skeletal structure, among other factors. However, our extensive research over many years has demonstrated that, in general, speech comprehension requires accurate identification of the first through fourth formants. It has been confirmed that the speech processing device 1 can achieve this effect by focusing on compensation in the 400 Hz to 5 kHz range.

[0056] 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.

[0057] According to the above configuration, the amplifier 14 amplifies the first audio signal SV1' but does not amplify the second audio signal SV2. Therefore, the audio processor 10 can output a synthesized audio signal in which the first-order formant components are not amplified and predetermined formant components (e.g., second-order formant components, third-order formant components, and fourth-order formant components) are amplified. Therefore, the audio processing device 1 can output a synthesized audio signal SSV that the listener perceives as clear without using an auxiliary device.

[0058] As a result, the possibility of misunderstandings occurring between the speaker and the listener or of the listener being offended can be reduced, making it easier for communication to proceed smoothly between the speaker and the listener.

[0059] The inventors have found that, when making audio clearer, it is preferable to perform component removal, amplification, and additive synthesis in this order. Component removal involves removing (1) low-frequency components and (2) high-frequency components from the first audio signal SV1. The low-frequency components are components with frequencies less than 400 Hz, and the high-frequency components are components with frequencies greater than 7 kHz. Amplification involves amplifying the first audio signal SV1' from which the low-frequency and high-frequency components have been removed. Additive synthesis is the additive synthesis of the first audio signal SV1" and the second audio signal SV2, which are in phase. However, the wavelength of a human voice is usually about 100 Hz or more and 1 kHz or less. Therefore, in many cases, it is sufficient for the filter 13 in the audio processor 10 that constitutes the audio processing device 1 to be able to remove only the low-frequency components from the first audio signal SV1. In other words, the filter 13 does not need to be configured to remove the high-frequency components from the first audio signal SV1. Therefore, the filter 13 can be configured simply, and costs can be reduced.

[0060] In the audio processing device 1, it is preferable that the phase corrector 12, the filter 13, the amplifier 14, and the additive combiner 16 are each configured by an analog circuit.

[0061] 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 using a digital circuit, the voice processing device 1 configured in this manner can output a synthesized voice signal SSV that the listener perceives as clearer. This is thought to be because the filter 13 configured using 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 using a digital circuit.

[0062] In the audio processing device 1 , the audio processor 10 preferably further includes an adjuster 15 .

[0063] According to the above configuration, the speaker can adjust the gain of the amplifier 14 depending on the state of the listener, regardless of the listener's intention. That is, the degree of clarity of the synthesized speech signal can be adjusted. Therefore, the speech processing device 1 configured in this manner can output a synthesized speech signal SSV with the degree of clarity adjusted depending on the listener.

[0064] 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-order formant components, third-order formant components, and fourth-order formant components.

[0065] 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. In the audio processing device 1 configured in this manner, the amplifier 14 amplifies the first audio signal SV1′ that falls within the frequency band that contributes to the clarity of an audio signal.

[0066] The audio processor 10 included in the audio processing device 1 is also an aspect of the present invention.

[0067] The voice processor 10 can output a synthesized 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.

[0068] In the audio processor 10, the audio signal SV and the synthesized audio signal SSV are preferably analog signals.

[0069] In the audio processor 10, the phase corrector 12, the filter 13, the amplifier 14, and the additive combiner 16 are preferably each configured by an analog circuit.

[0070] This allows for easy configuration and miniaturization of the audio processor 10. Furthermore, the audio processor 10 configured in this manner can output a synthesized audio signal that the listener perceives as clearer than when the filter 13 is configured using a digital circuit.

[0071] [Embodiment 2] Another embodiment of the present invention will be described below. For ease of explanation, the same reference numerals will be used to designate components having the same functions as those described in the above embodiment, and the description thereof will not be repeated.

[0072] <Configuration of Audio Processing Device> As shown in FIG. 4, the audio processing device 1A includes a microphone 21, a speaker 22, and a housing similar to those of the first embodiment, as well as an audio processor 10A.

[0073] [Configuration of Audio Processing Device] The audio processor 10A includes an audio signal generation unit 11A, a correction signal generation unit 12A, a filter 13A, and a synthesis unit 14A. The audio processor 10A according to this embodiment further includes a first port 15A, an input buffer unit 16A, an adjustment unit 17, and a second port 18. Note that the audio processor 10A may further include a power supply unit, a detection unit, and a control unit, all of which are not shown.

[0074] (First Port) The first port 15A outputs an audio signal x representing audio 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-order formant component h1, a second-order formant component h2, ..., and an nth-order formant component hn. Note that n depends on the user on the transmitting side and, although there may be some individual differences, is a positive integer of at least 4 or greater. Note that the first-order formant component h1, the second-order formant component h2, ..., and the nth-order formant component hn are described in Patent Document 1, and therefore will not be described in this embodiment.

[0075] (Input Buffer Unit) As shown in FIG. 4 , the input buffer unit 16A is provided between the first port 15A and the audio signal generator 11A. The input buffer unit 16A can be configured with a microphone amplifier, a preamplifier that lowers output impedance, or the like, depending on the usage situation. The input buffer unit 16A according to this embodiment adjusts the audio signal x by narrowing the upper and lower frequency limits and increasing the sound pressure, as shown in FIG. 6 , for example. Then, as shown in FIG. 4 , the input buffer unit 16A outputs the adjusted audio signal x to the audio signal generator 11A. Note that if the audio signal supplied from the first port does not require adjustment (i.e., if 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.

[0076] (Audio Signal Generator) The audio signal generator 11A generates a first audio signal x1 and a second audio signal x2 having identical spectra from the audio signal x supplied from the input buffer 16A. "Having identical spectra" 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 generator 11A according to this embodiment also generates a third audio signal x3 having an identical spectrum to the second audio signal x2. The audio signal generator 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 generator 12A. The audio signal generator 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 generation 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, i.e., so that 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 generation unit 11A may also duplicate the audio signals supplied from the input buffer unit 16A, and use the original audio signal as the first audio signal x1, and the duplicated audio signals as the second audio signal x2 and the third audio signal x3.

[0077] (Correction Signal Generator) The correction signal generator 12A generates a phase correction signal p based on the second audio signal x2 supplied from the audio signal generator 11A. As described above, the audio signal generator 11A according to this embodiment generates the third audio signal x3 whose phase is equal to that of the second audio signal x2. Therefore, the correction signal generator 12A according to this embodiment generates the phase correction signal p based on the third audio signal x3. As a result, the correction signal generator 12A generates the 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. The correction signal generator 12A then outputs the generated phase correction signal p to the operational amplifier 132a of the filter 13A.

[0078] As shown in FIG. 7 , the correction signal generator 12A according to this embodiment includes a capacitor 121 and a resistor 122. One end of the capacitor 121 is connected to the audio signal generator 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 generator 12A configured in this manner outputs a phase correction signal p containing only frequency components equal to or greater than ½πCR (C: capacitance, R: resistance) using a high-pass filter function internally formed by the capacitor 121 and the resistor 122. Hereinafter, the lower limit of the frequency components contained in the phase correction signal p will be referred to as the phase correction start frequency fs. The correction signal generator 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 in detail later, 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 first audio signal x1 and the second audio signal x2 after correction.

[0079] 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 generator 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 generator 12A can be configured with a simple configuration that is compatible with audio signals of various frequencies.

[0080] (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 the upper limit frequency of the first formant component h1 and the lower limit frequency of the 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 that is 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.

[0081] The high-pass filter 131 removes a first frequency component from the first audio signal x1 supplied from the audio signal generation unit 11A. In the high-pass filter 131 according to this embodiment, the first frequency is set to 400 Hz. The high-pass filter 131 then outputs the first audio signal x1' from which the first frequency component has been removed to the low-pass filter 132. The first audio signal x1' is an audio signal that does not include a first-order formant component h1, but does include a second-order formant component h2, a third-order formant component h3, ..., an n-th order formant component hn.

[0082] 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, ..., the n-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 boundary frequency at which components are removed or not can be set arbitrarily depending on the intended use.

[0083] 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, supplied from the high-pass filter 131. The second frequency component is a frequency component equal to or greater than the second frequency. The second frequency (cutoff frequency) is a frequency between the upper limit frequency of the fifth-order formant component h5 and the lower limit frequency of the sixth-order formant component h6, or a frequency between the upper limit frequency of the sixth-order formant component h6 and the lower limit frequency of the seventh-order 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, or higher frequency components have been removed, i.e., a speech signal containing only second- to fifth-order formant components h2 to h5, or only second- to sixth-order formant components h2 to 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 synthesized speech signal x', which will be described later.

[0084] However, when frequency components are removed from the first speech signal, a phase delay inevitably occurs. If a first speech signal x1, whose phase is delayed by time T, is combined with a second speech signal x2 without correction, the voltage V1 of the first speech signal x1 becomes −V1 due to the phase delay, as shown in FIG. 9 . Therefore, the voltage V of the combined speech signal becomes V2 + (−V1) because the first speech signal x1 and the second speech 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, causing extremely low sound pressures for the first-order formant component h1 and the second-order formant component h2, which are particularly important for speech comprehension, and other problems that significantly hinder improvements in clarity.

[0085] For this reason, the filter 13A has a function of correcting 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 also configured so that the first audio signal is input to its inverting input terminal (−). The operational amplifier 132a receives a phase correction signal p at its non-inverting input terminal (+), and outputs a first audio signal x1′ whose phase has been corrected to approach the phase of the second audio signal x2. Specifically, the operational amplifier 132a compares 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 a first audio signal x1′ whose phase has been advanced in accordance with the voltage difference. On the other hand, if the voltage of the phase correction signal p is lower, the operational amplifier 132a outputs a first audio signal x1' whose phase has been delayed in accordance with the voltage difference. As described above, the voltage of the phase correction signal p varies in accordance with the phase correction start frequency fs determined by the capacitor 121 and resistor 122 of the correction signal generation 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, a phase correction signal p having a voltage corresponding to the adjusted phase correction start frequency fs is input to the non-inverting input terminal (+) of the operational amplifier 132a. The operational amplifier 132a then outputs a first audio signal x1' corrected so that the difference in phase with the second audio signal x2 is minimized.

[0086] The filter 13A may be configured such that the low-pass filter 132 first removes the second frequency component from the first audio signal x1. Then, 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.

[0087] (Adjustment Unit) The adjustment unit 17 adjusts the sound pressure of the phase-corrected first audio signal x1'. 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 the 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 comprehension, to a level that exceeds the masking line MK', which is an audible level. Furthermore, the adjustment unit 17 can freely select the output level curve of the synthesized audio signal x' to be "minimum," "maximum," or "intermediate," as shown in FIG. 11 .

[0088] (Synthesis Unit) The synthesis unit 14A generates a synthesized 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 whose 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 this embodiment includes the adjustment unit 17. Therefore, the synthesis unit 14A according to this embodiment additively synthesizes the first audio signal x1, from which the sound pressure has been adjusted, and the second audio signal x2. The synthesis unit 14A then supplies the generated synthesized 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 uncorrected first audio signal x1 by T'. Therefore, the corrected first audio signal x1' is in phase with the second audio signal x2. As a result, the frequency characteristics of the synthesized audio signal x' output by the synthesis unit 14A are as shown in FIG. 13.

[0089] (Second Port) The second port 18 outputs the synthesized voice signal x′ supplied from the synthesis unit 14 A to a device connected to the second port 18 .

[0090] (Power Supply Unit) The power supply unit supplies power to each unit requiring power in the audio processor 10 A. The form of the power supply unit is not limited, and may be an AC / DC converter such as an AC adapter, or a battery.

[0091] (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 it is not functioning normally is a power outage. Furthermore, if the power supply unit is a battery, an example of a state in which it is not functioning normally is a dead battery.

[0092] (Selection Unit) The selection unit selects whether or not to apply audio processing to the audio signal x based on control by 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 and 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, where it is synthesized with the second audio signal x2. 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.

[0093] (Control unit) The control unit controls the selection unit to perform or not perform audio processing on the audio signal x according to the detection result of the detection unit. If the detection result of the detection unit is "Yes", the control unit controls the selection unit to perform audio processing on the audio signal x (turn ON). On the other hand, if the detection result of the detection unit is "No", the control unit controls the selection unit not to perform audio processing on the audio signal x (turn OFF).

[0094] [Modification of Audio Processing Device] In the audio processor 10A in the above embodiment, the high-pass filter 131, the low-pass filter 132, the correction signal generator 12A, and the combiner 14A are configured as analog circuits. However, at least one of these may be configured as a digital circuit.

[0095] Furthermore, 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.

[0096] 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.

[0097] The audio processor 10A may further include a selector, an amplifier, a circuit separator, a transmitter output adjuster, and a polarity switcher interposed between the combiner 14A and the second port 18.

[0098] The audio processor 10A may also include an attenuator between the synthesis unit 14A and the second port 18. The attenuator arbitrarily adjusts the output to match the input audio source signal. By incorporating such an attenuator into the signal line of a microphone in an existing device that transmits audio via wired or wireless communication, or into a television, radio, or other entertainment playback device, it becomes possible to provide a system that accommodates hearing loss symptoms due to hearing impairment.

[0099] [Operation and Effect of Audio Processing Device] The audio processor 10A described above 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 audio even clearer than before.

[0100] [Modifications] The voice processors 10, 10A may be realized to include or be combined with the configurations described in the patent publications associated with the following patent numbers, for example. The technologies described in Japanese Patent Nos. 3731179 and 6548938 are technologies that perform voice processing for voice clarity by focusing on first-order (or lower-order) formant components and higher-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 FIGS. 2, 4, and 6 of Japanese Patent No. 3731179 or the configuration shown in FIG. 5 of Japanese Patent No. 6548938 may be adopted to realize the voice processor 10. The audio processor 10 is required only to perform audio processing on the audio signal representing the voice of the speaking party supplied from the microphone 21 to clarify the audio, and to supply the processed audio signal to the speaker 22.

[0101] Moreover, instead of the audio processor 10 shown in FIG. 5, the configuration shown in FIG. 4 of Japanese Patent Publication No. 7105756 can be adopted.

[0102] [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.

[0103] [Summary] The information processing device according to the first aspect of the present invention is a portable audio processing device, and includes an audio processor that performs audio processing on an audio signal supplied from a microphone to make the audio clearer, a speaker that converts the processed audio signal into audio, and a housing that houses at least the audio processor.

[0104] According to the above configuration, the voice processing device is configured to be portable, and therefore it is possible to provide a voice processing device that can be used in a state where it can be moved along with the speaker.

[0105] Because the speaker and the voice processing device can be moved together, the voice processing device can be used in a variety of situations, such as when a store clerk interacts with customers, when an event venue attendant 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.

[0106] 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 may be an unspecified person or people such as passersby.

[0107] In addition, in the audio processing device according to the 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.

[0108] According to the above configuration, the voice processor and the speaker are housed integrally in a single housing, making it easy for the speaker to carry the voice processor.

[0109] In addition, in the audio processing device according to the 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 is configured to accommodate the microphone in addition to the audio processor and the speaker.

[0110] According to the above configuration, the voice processor, the speaker, and the microphone are housed integrally in a single housing, making it easy for the speaker to carry the voice processor. An example of such a voice processor is a device called an electronic megaphone or loudspeaker.

[0111] Furthermore, 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 in which the device further includes a microphone that converts audio into an audio signal.

[0112] The information processing device according to one aspect of the present invention can also be sold as a package including a microphone.

[0113] 1 Audio processing device 10 Audio processor 11 Branching section 12 Phase corrector 13 Filter (band-pass filter) 14 Amplifier 15 Adjuster 16 Additive synthesizer P11 Port (first port) P12 Port (second port) 21 Microphone 22 Speaker 1A Audio processing device 10A Audio processor 10, 10A Audio processor 11A Audio signal generation section 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 First port 16A Input buffer section 17 Adjustment section 18 Second port 21 Microphone 22 Speaker 24 Support member

Claims

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

2. The audio processing device according to claim 1, wherein the housing accommodates the speaker in addition to the audio processor.

3. The audio processing device according to claim 2, further comprising a microphone that converts audio into an audio signal, wherein the housing accommodates the microphone in addition to the audio processor and the speaker.

4. The audio processing device according to claim 1 or 2, further comprising a microphone for converting audio into an audio signal.

Images