JP2016110050A - Voice processor, voice clearing device, and voice processing method - Google Patents

Abstract

PROBLEM TO BE SOLVED: To provide a novel voice processor, a voice clearing device, and a voice processing method which make it easy to hear voice from speakers, even in an existing acoustic device and a broadcasting facility.SOLUTION: Collected audio signals are branched, a low frequency component equal to low-order formant or lower and a high frequency component beyond high-order formant are removed from one of the branched audio signals, a phase of the other audio signal is synthesized with an audio signal corrected so that it matches a phase of the processed audio signal and processed, and outputted. Thus, it is possible to extract from the original audio signal only formant components of second to fourth degree significance for language discrimination, to add it to an original audio signal, and to output it, so that contents of the voice can be clearly heard.SELECTED DRAWING: Figure 5

Description

  The present invention relates to an audio processing device, an audio clarifying device, and an audio processing method for making it easy to clearly hear audio flowing from a speaker such as an audio device or broadcasting equipment.

  In general, broadcasting facilities (in-house broadcasting) are installed at places where a large number of people gather, such as buildings, public building halls, shopping centers, sports facilities, etc., for guiding necessary information and making emergency announcements. In this broadcasting equipment, for example, speakers are installed on the ceiling or wall of a building, and the audio signal sent from the broadcasting room is converted into audio by the speakers, but the resonance phenomenon of the ceiling plate and other speakers Depending on the conditions such as interference with the voice coming out, the contents of the voice may be difficult to hear.

  On the other hand, in the case of an elderly person or a hearing-impaired person whose hearing ability has declined due to aging, even if the voice can be heard, it may be difficult to understand what is being said. This is thought to be because the speech (language) produced by human beings consists of vowels and consonants, and among these vowels, it is difficult to hear consonants even if they can be heard relatively easily. Therefore, simply raising the volume of the sound from the speaker or increasing the sensitivity of the hearing aid increases not only the consonant but also the vowel, which cannot be easily eliminated.

  For this reason, for example, in Patent Document 1 below, audio that corrects frequency characteristics and levels of audio output from a receiver or the like in accordance with changes in hearing depending on the age of a user who uses an audio device such as a mobile phone, a television, or a stereo. Correction devices have been proposed. In Patent Documents 2 and 3 below, a method of detecting a formant of a speaker and transforming it into a formant that is easy to listen to is proposed.

JP 2000-209698 A JP 2008-116534 A Japanese Patent Laid-Open No. 01-93796

  By the way, in order to make it easier to hear the sound from the speakers in the in-house broadcasting as described above, it is possible to replace the broadcasting equipment and speakers with high-quality ones, devise the number of speakers and the installation location, etc. This requires a great deal of money. On the other hand, in the sound correction method for the elderly or hearing-impaired people whose hearing ability has declined, the voice color (sound quality) may change greatly, resulting in an unnatural sound.

  Therefore, the present invention has been devised to solve these problems, and one of the purposes is to make it easier to hear the sound flowing from the speaker even in an existing audio device or broadcasting facility. A novel speech processing device, speech clarification device, and speech processing method are provided. Another object of the present invention is to provide a novel speech processing device, speech clarification device, and speech processing method that can be listened to with natural speech without any sense of incongruity even if the hearing ability is aged or deaf. It is to provide.

  The understanding of language by human hearing is that sound waves received by the eardrum are converted into electrical signals by the auditory organ and transmitted to the brain, and are judged as words by collating with memories obtained from past experiences. The auditory organ is very complex and its function is not well understood.

  In recent years, passive devices such as hearing aids and sound collectors with special auxiliary sound processing for “individual” for hearing-impaired people are becoming widespread. There are many cases that hesitate to use because of the tiredness and appearance due to wearing. Moreover, the higher the performance, the higher the cost and the greater the economic burden.

  On the other hand, according to the research on the utterance mechanism of language, air whose strength, amount, etc. are adjusted is emitted as a vowel from an organ called a vocal cord. This is called the first formant, and it has resonance energy by adjusting the following trajectory, oral volume and shape, nasal cavity, tongue vibration and shape, maxillary and mandibular relaxation volume, and so on. A plurality of peaks on the frequency characteristic appear, and these are called formants. Peaks that appear in order from the lower frequency side toward the higher frequency side are the second formant, the third formant, the fourth formant,..., The nth formant, and a synthesized voice is emitted as a voice unique to the individual.

  In general, auditory comprehension of speech signals is realized by detecting heavy frequency components and their magnitude in the voice flow and transmitting them to the brain. The field is relatively advanced, and the first to fourth formants are particularly heavy frequency components.

  Accordingly, a first invention for solving the above-mentioned problem branches a collected audio signal, and a low frequency component lower than a low-order formant and a high frequency exceeding a high-order formant from one of the branched audio signals. A band-pass filter that removes components, a phase corrector that corrects the phase of the other branched audio signal so as to match the phase of the audio signal that has passed through the band-pass filter, and the band-pass filter that has passed An audio processing apparatus comprising: an adder / synthesizer that adds an audio signal to the audio signal that has passed through the phase corrector and outputs the resultant signal.

  According to such a configuration, only the second to fourth formant components important for language identification can be extracted from the original voice signal and output in addition to the original voice signal. By emphasizing the second to fourth formant components attenuated due to worsening conditions, the sound can be clearly heard even in broadcasting facilities with poor conditions. On the other hand, especially for subjects who are difficult to hear the second to fourth formant components, such as elderly people and hearing-impaired people whose hearing has declined, the contents of the sound must be clearly heard without increasing the overall volume. Can do.

  Here, although it is not particularly limited as a frequency band according to the present invention, “a low-frequency component below a low-order formant and a high-frequency component exceeding a high-order formant are removed”, it is important for language recognition, for example. This is a frequency band including the second to fourth formants, more preferably the second to fifth formants (hereinafter, the same).

  According to a second aspect of the present invention, there is provided a bandpass filter for removing a low-frequency component below a low-order formant and a high-frequency component exceeding a high-order formant from the left and right audio signals to be collected stereo signals, and the left and right audio signals A pair of phase correctors that respectively correct the phase of the audio signal that has passed through the bandpass filter, and the left and right audio signals that have passed through the bandpass filter as the audio signal that has passed through the bandpass filter. And a pair of adder / synthesizers that respectively add and synthesize and output. According to such a configuration, even in the case of stereo sound, it is possible to output clear and easy-to-listen sound as in the first invention.

  According to a third invention, in the first or second invention, the bandpass filter passes a high-pass filter that removes a low-frequency component below a low-order formant from one of the branched audio signals, and the high-pass filter. An audio processing apparatus comprising: a low-pass filter that removes a high-frequency component exceeding a high-order formant from an audio signal. According to such a configuration, only the second to fourth formant components important for language identification can be reliably extracted from the original speech signal.

  According to a fourth aspect of the present invention, there is provided the speech processing apparatus according to any one of the first to third aspects, further comprising an effect adjuster for adjusting a value of the speech signal before synthesis that has passed through the bandpass filter. Depending on conditions, performance of broadcasting facilities, etc., attenuation amounts of the second to fourth formant components important for voice identification are not uniform. In addition, in the case of an elderly person or a hearing-impaired person whose hearing ability has declined, there is a great individual difference in the ability to listen to formant components of the second to fourth degree. Therefore, by adjusting (increasing / decreasing) the values of the second to fourth formant components extracted by the bandpass filter by the effect adjuster as in the present invention, clear and uncomfortable voice can be output. .

  According to a fifth aspect of the present invention, any one of the first to fourth sound processing devices is housed in a portable housing, and a microphone for collecting sound is detachably connected to the surface of the housing. The speech clarification device includes an input side connection port and an output side connection port that outputs an audio signal processed by the audio processing device to broadcasting equipment. According to such a configuration, the apparatus of the present invention can be easily incorporated (attached) to existing broadcasting equipment, so that an excellent effect can be exhibited at low cost. In addition, since it is easy to carry, it can be easily applied to broadcasting facilities at outdoor event venues and other broadcasting facilities.

  According to a sixth aspect of the present invention, there is provided a first step of branching a collected sound signal, a low frequency component equal to or lower than the first formant and a high frequency exceeding a higher order formant from one of the sound signals branched in the first step. A second step of removing frequency components, and a third step of correcting the phase of the other audio signal branched in the first step so as to match the phase of the audio signal processed in the second step. A voice processing method comprising: a step; and a fourth step of synthesizing and outputting the voice signal processed in the second step and the voice signal processed in the third step. . According to such a configuration, it is possible to output a clear and easy-to-listen sound as in the first invention.

  According to the present invention, only the second to fourth formant components important for language identification can be extracted from the original audio signal and output in addition to the original audio signal. By emphasizing the second to fourth formant components attenuated by, the sound can be clearly heard even in poorly conditioned broadcasting facilities. On the other hand, especially for subjects who are difficult to hear the second to fourth formant components, such as elderly people and hearing-impaired people whose hearing has declined, the contents of the sound must be clearly heard without increasing the overall volume. Can do.

It is the figure which showed the example of a frequency analysis seen in the flow of an audio | voice. It is the figure which showed the example explaining the process of the language recognition sampled from the flow of an audio | voice. It is the figure which showed the example which increased the audio | voice level of V in sound-sensitive deafness by 10 dB. It is the figure which showed the state which extracted the formant of G area | regions other than the 1st formant h1 with the bandpass filter, and added and synthesize | combined each harmonic component signal of each formant which carried out amplitude amplification to the main audio | voice signal. It is the figure which showed one Embodiment of the audio processing apparatus 100 which concerns on this invention. 3 is a circuit configuration diagram of a phase corrector 3. FIG. It is a frequency characteristic figure of input buffer 2 and bandpass filter 30 (4, 5). It is a phase shift characteristic figure by bandpass filter 30 (4, 5). It is a figure which shows the phase difference T when there is no phase corrector. It is a frequency characteristic figure by phase difference. FIG. 4 is a diagram showing a phase variable range of the phase converter 3. It is a figure which shows the output characteristic by the effect regulator. It is a figure which shows the same phase addition in case there exists the phase corrector 3. FIG. It is a figure which shows the structure of the existing cable broadcasting equipment. It is a figure which shows the structure of the existing radio broadcasting equipment. It is the figure which showed one Embodiment of the speech clarification apparatus 200 which concerns on this invention. It is the figure which showed other embodiment (stereo sound) of the audio processing apparatus 100 which concerns on this invention.

  Hereinafter, embodiments of the present invention will be described with reference to the accompanying drawings. FIG. 1 shows an example of frequency analysis observed in a voice flow. As described above, when a person utters a word, several harmonics called formants are emitted simultaneously with the resonance phenomenon. Of these several harmonics, the fundamental wave h1 that mainly affects the vowel is called the first formant, the second highest form harmonic h2 is the second formant, and the second highest harmonic h3 is the second formant. The third formant, and then the harmonic h4 with the next highest frequency is called the fourth formant, and the harmonic hn with the higher frequency in turn is called the nth formant. The higher the formant, the lower the level, but it has been found that the second to fifth formant components are particularly important for language understanding.

  According to research in the language field, hearing is sensitive to detecting each formant and the ability to perceive words is learned and remembered in early childhood. For this reason, even if a part of the language information obtained from hearing is missing, the missing part is complemented by the experience stored and accumulated in the past, and it is said that there is no problem in general daily conversation.

  However, in order to recognize the sound that has been heard as words, it is necessary for the auditory sense to detect the formants, but if the important formant components of these formants cannot be detected, they cannot be recognized as words. For example, an experimental result has been reported that the language cannot be understood at all if the band of 500 Hz to 3 kHz is deleted.

  There are three types of hearing loss that are difficult to hear: hearing loss due to disturbance of the outer ear and middle ear, sensory hearing loss due to damage to the inner ear, auditory nerve, and brain, and complex hearing loss due to both. Broadly classified. Of these, in the case of conductive hearing loss, which is mildly deaf, it can be dealt with, for example, by increasing the volume of the TV or by talking to the speaker in the ear, but in the case of sensory hearing loss, it progresses with aging. Even if the incoming sound is made louder, I can't understand the content of the word just by hearing it as a “gohon, gohon” sound. This is presumably because the ability to detect medium level formants from the second formant to the fourth formant was reduced even when the volume was increased.

  In addition, hearing has the property that when an excessive sound reaches, other small sounds near the frequency of the sound are difficult to detect, or one sound is disturbed by another sound and difficult to hear. This is called the masking effect and is said to be due to the brain's self-defense action against excessive sounds. As a representative example of these, for example, in a broadcasting facility installed in a large venue, the sound pressure level in the low range included in the sound component emitted from the speaker embedded in the ceiling is about 2 to 3 times due to the baffle effect by the ceiling board. May rise.

  As a result, among the audio components output from the speaker, a formant component equal to or higher than the second formant is masked, so that the broadcast content cannot often be heard. On the other hand, in the case of an elderly person, the voice detection ability is reduced, so that when the volume of the first formant, which is the fundamental wave of the language, is increased, the formants after the second formant are masked and the words cannot be heard.

  From the above considerations, in order to accurately convey the language, it is considered that these deafness problems can be solved by amplifying higher harmonic components than the second formant to exceed the masking level.

  FIG. 2 shows an example illustrating a language recognition process sampled from a speech flow. The horizontal axis represents frequency (Hz) and the vertical axis represents sound pressure (dB), which indicates the formant component included in the voice V (Vu) and its level. The first formant h1 has the lowest frequency component (vowel), but its sound pressure is at the highest level. Thereafter, as the second formant h2 to the nth formant hn, the sound pressure gradually changes to a low level.

  In this figure, a line MK indicates a masking range for a normal hearing person, and a line MK ′ indicates a masking range for a hearing impaired person. As shown in the figure, the masking line MK of the normal hearing person is lower than all the formants, so that the V voice can be clearly heard without being affected by the masking effect, whereas the masking line MK ′ of the hearing person is other than the first formant h1. It can be seen that the V sound cannot be heard because the masking effect is remarkably high.

  FIG. 3 shows a case where the sound level of V in sound-sensitive hearing loss is increased by 10 dB. In the figure, h1 + to hn + indicate levels obtained by increasing each formant h1 to hn by 10 dB, respectively, but the deaf person's masking line MK ′ is simply moved upward in parallel, and simply increasing the volume It turns out that language understanding is impossible.

  FIG. 4 shows that a G region formant (second formant h2 to nth formant hn), which is an important element for speech recognition, is extracted by a bandpass filter (bandpass filter) and amplified in amplitude. The harmonic component signal formants h2 + to hn ++ are added and synthesized to the main audio signal. As shown in the figure, it can be understood that if such processing is performed, the voice V can be understood because all formant components exceed the masking line MK ′ of the hearing impaired.

  Note that conventional hearing aids and sound collectors mainly collect sound generated from a distant location, but higher-order formant components in the middle of propagating in the air are ambient noise, diffusion, and air mass. As a result, the language comprehension is significantly hindered, so that more complicated speech processing is necessary.

  The speech processing apparatus of the present invention has less propagation loss due to speech picked up by a microphone at a distance close to the mouth of the speaker, that is, air, etc., and attenuates higher-order formant components that are important for language understanding. If the main target is pure speech with no sound, and the higher-order formant components are extracted from the speech and added to the original speech, the content of the speech is clearly seen even in poorly conditioned broadcasting facilities and deaf people. It was devised based on the knowledge that it is possible to listen to music.

  FIG. 5 shows an embodiment of the speech processing apparatus 100 according to the present invention. In the figure, reference numeral 1 is a microphone for sound collection, 2 is a buffer amplifier for amplifying a minute voltage signal uttered by the microphone 1 several times, and 3 is an audio signal amplified by the buffer amplifier 2. Similarly, a phase corrector 30 corrects the phase of the sound signal, and is a bandpass filter 30 that removes a low-order formant frequency component and a high-order formant frequency component from the audio signal amplified by the buffer amplifier 2. .

  In the figure, 6 is an adjuster that adjusts the value of the audio signal that has passed through the bandpass filter 30, and 7 is an adder that combines the audio signal adjusted by the adjuster 7 with the audio signal corrected by the phase corrector 3. 8 is an output buffer for adjusting the output of the audio signal synthesized by the adder / synthesizer, 9 is an attenuator for adjusting the audio signal to the output sensitivity of the microphone 1, and 10 is an output terminal. .

  The band-pass filter 30 further includes a high-pass filter 4 and a low-pass filter 5. The high-pass filter 4 removes low-frequency components below the first formant h 1 from the audio signal, and the low-pass filter 5 removes the high-pass filter 4. The high-frequency components exceeding the higher-order formants are removed from the audio signal that has passed through, and the second to fifth formant frequency components are extracted.

  As shown in FIG. 6, the phase corrector 3 has a circuit configuration including an input terminal 15, an input ground terminal 16, an output terminal 17, an output ground terminal 18, an operational amplifier 19, a capacitor, resistors R, R1, and R2. The audio signal can be arbitrarily controlled in phase delay in the range of 0 to -180 degrees as will be described later.

  Hereinafter, the operation of the speech processing apparatus 100 having such a configuration will be described. As shown in FIG. 5, firstly, the voice uttered by the speaker's mouth is collected directly by the microphone 1 without being attenuated, converted into an electric signal (speech signal), and amplified several times by the buffer amplifier 2. . The amplified audio signal is then branched, one of which passes through the main audio path L1, is sent to the phase corrector 3, and the phase thereof is corrected as will be described later.

  On the other hand, the frequency components of the second formant to the fifth formant are extracted from the sound signal sent to the sub sound path L2 side by the bandpass filter 30. That is, by passing through the high pass filter 4 of the bandpass filter 30, the first formant and harmful low frequency components, for example, wind noise generated by the breath blown to the microphone 1, is suppressed, and then passes through the low pass filter-5. This suppresses annoying high-frequency noise generated by the amplifying element itself of each part, thereby providing a kind of broadband filter characteristic indicated by a curve na in FIG. 7, thereby improving the improvement effect.

  The high-pass filter 4 also has a function as an amplifier of about 5 times to arbitrarily set the degree of intelligibility, and the low-pass filter-5 further includes a wide frequency component including a super high-order formant having a low contribution to language recognition. Eliminating the signal-to-noise ratio performance characteristics. The frequency characteristic of the output point z of the band filter 30 is shown by a curve bp in FIG. In the present embodiment, the frequency characteristic of the band filter 30 is about 4 octaves from 400 Hz to 7 kHz set in various verification tests, and both low and high frequencies are cut off at about −12 dB / oct. Yes.

  The value of the audio signal that has exited the band filter 30 is adjusted by the effect adjuster 6 and then added and synthesized by the adder / synthesizer 7 to the audio signal on the main audio path L1 side. The added and synthesized audio signal is appropriately adjusted by the output buffer 8 and the attenuator 9 and then output from the output terminal 10 to existing broadcasting equipment.

  Here, in the band filter 30, the essential characteristic is that the input signal from the point x in FIG. 5 is output to the point z with a phase delay T of about 90 degrees at the maximum in the voice band as shown in FIG. There is. For this reason, when the audio signal output at the z point is directly synthesized with the main audio signal passing through the main audio path L1 by the adder / synthesizer 7, the main audio signal voltage is set to Vx, as shown in FIG. Assuming that the output voltage of 30 is Vz, Vz is considered to be −Vz because of the phase lag, so Vx + (− Vz) = Vk, and the voltage difference between the two signals becomes the output of the adder / synthesizer 7.

  As a result, as shown in FIG. 10, a dead zone called “deep valley” characteristic, which is d in the frequency characteristic, is generated in a certain frequency region, and the volume of the second formant and the third formant gold bread, which is particularly important for language understanding, is extremely high. Inconveniences that become a major obstacle to improvement in clarity such as reduction in size occur. FIG. 9 shows the signal cancellation phenomenon in the adder / synthesizer 7 when the phases of the two audio signals are taken into consideration.

  In the present invention, in order to eliminate such a dead zone, a phase corrector 3 is provided on the main audio signal path L1, and a bandpass filter 30 (4, 5) for higher-order formant extraction as shown in FIG. The phase delay corresponding to the phase delay generated in step 1 is added and corrected, and the generation of the dead zone is avoided by adding and synthesizing the corrected audio signal with the audio signal passed through the bandpass filter 30 (4, 5). It is.

  Therefore, the phase corrector 3 provided on the main voice path L1 side plays an important role in realizing the voice processing apparatus 100 of the present invention and is a major feature. That is, in the circuit constituting the phase corrector 3 as shown in FIG. 6, the angular frequency is determined by ω = 1 / CR, and phase delay control can be arbitrarily achieved in the range of 0 to −180 degrees. Incidentally, by arbitrarily selecting C and R values, there is an excellent characteristic that the phase delay can be as close to 0 degrees as possible even in a low frequency band while keeping the gain constant. Shown in

  In FIG. 6, since the audio signal input from the input terminal 15 is applied to the inverting terminal (−) of the operational amplifier 19, an inverted signal having a phase opposite to that of the input signal appears at the output terminal 17. On the other hand, at the same time, the capacitor C is connected to the input terminal 15 to the non-inverting terminal (+) of the operational amplifier 19, but the capacitor C is an element whose impedance changes with frequency and its impedance Z is 1 / ωC. Therefore, the internal resistance (impedance) decreases as the frequency increases.

  Therefore, the higher the frequency, the more the input signal to the non-inverting input terminal increases. Both signals are calculated inside the operational amplifier 19, and if the non-inverting side signal is large, the phase delay width of the signal at the output terminal 17 is reduced. Since the impedance increases as the frequency decreases, the input signal to the non-inverting input terminal decreases, and a signal having a large delay width is output to the output terminal 17, but the degree of phase delay at the desired frequency Can be freely set by the constants of the resistor R and the capacitor C. The amplification factor of the phase corrector 3 is determined by r2 / r1, and can be arbitrarily set according to the purpose, and can always be kept constant.

  Then, a high-order formant component important for language understanding in the G range of FIG. 4 is applied to the adder / synthesizer by the effect adjuster 6 and arbitrarily set so as to exceed the masking line MK ′. At the composite output point t, as shown in FIGS. 12A to 12C, the frequency versus output level curve has “minimum position”, “center position”, and “maximum position” depending on the setting position of the effect adjuster 6. Thus, it can be freely selected so as to obtain an optimum effect according to the place of use and the degree of hearing loss.

  Furthermore, since the sound processing apparatus 100 of the present invention is provided with the attenuator 9 on the output side, the output sensitivity can be matched with the output sensitivity of the microphone 1, so that the existing sound processing apparatus as shown in FIGS. It can also be easily incorporated into wired and wireless broadcasting facilities. As a result, it is not necessary to modify or improve the existing broadcasting equipment, and an excellent effect of improving the sound quality of the in-house broadcasting can be obtained at low cost.

  In this case, as shown in FIG. 16, the voice processing device 100 of the present invention is built in a portable (portable) casing 33 (metal case), and the surface of the casing 33 is used for collecting sound. A unit-like audio clarifying device configured to include an input-side connection port 34 for detachably connecting the microphone 1 and an output-side connection port 35 for outputting an audio signal processed by the audio processing device 100 to broadcasting equipment. If it is 200, not only attachment and detachment will be easy, but also it can be easily carried and accommodated, so it can be easily applied to broadcasting equipment at outdoor event venues and other broadcasting equipment.

  As another embodiment of the sound processing apparatus 100 according to the present invention, a pair of main sound paths L1 are provided as shown in FIG. 17, and the sound signals of the main sound paths L1-R and L1-L are described above. Similarly, band-filtering is performed on the sub-main audio path L2, and the extracted signals (second formant to fifth formant) are output to the audio signals of the main audio paths L1-R and L1-L by the respective adder / synthesizers 7a and 7b. The same effect can be obtained even in the case of stereo sound.

  That is, as shown in FIG. 17, the left and right audio signals, which are stereo signals input from the pair of signal input terminals 1a and 1b, are amplified by the input buffer devices 2a and 2b, and the subsequent operation is made normal. Therefore, low impedance output is converted. The left and right audio signals (stereo signals) from the input buffer devices 2a and 2b branch to the main audio paths L1-R and L1-L and the input adder 31 side, respectively.

  The left and right audio signals (stereo signals) branched to the input adder 31 side are merged here, converted to a monaural signal by the mixing buffer 32, and amplified by the bandpass filter 30 including the high-pass filter 4 and the low-pass filter 5. At the same time, a high-order formant is extracted and sent to the effect adjuster 6 to adjust its value.

  On the other hand, the left and right audio signals (stereo signals) passing through the main audio paths L1-R and L1-L are inevitably in phase with the bandpass filter 30 as described above with the phase correctors 3a and 3b. The time delay corresponding to the generated phase delay is corrected to have the same phase, and are added and synthesized with the monaural audio signal whose effect is adjusted by the effect adjuster 6 in the adder / synthesizers 7a and 7b. Thereafter, the left and right audio signals (stereo signals) are output from the signal output terminals 10a and 10b after the impedance is lowered by the respective output buffers 8a and 8b.

  In general, audio of a television broadcast is broadcast in monaural, and the spread component is separated into left and right channels to create a sound field. Therefore, the output of the input adder 31 and the mixing buffer 32, that is, the sound component is larger than the spread component. It is applied to the bandpass filter 30 at a level about 6 dB larger. Therefore, clarification can be achieved by synthesizing the second formant to the fifth formant, which are important for language recognition, from the speech components and synthesizing them into the original stereo signal.

  As a result, it is possible to listen clearly even to elderly people and hearing-impaired people whose hearing has declined by incorporating sound signals that are easy to hear and clarified into wireless transmission devices, TV sound auxiliary devices, music playback devices, etc. It can also be used in daily life in a noisy environment. And when the device 100 of the present invention is actually incorporated into an existing wireless transmission device, TV audio assist device, music playback device, etc., it has become possible to eliminate the obstacles that occur in the daily life of the normal hearing person and the deaf person. Field test results are also reported.

  The voice processing apparatus 100 and the voice clarification apparatus 200 that can be easily carried can be applied to almost any existing broadcasting equipment, and can easily exhibit an excellent voice clarification effect. For example, call facilities such as hospitals, local government emergency broadcast facilities, telephone transmitters, wireless communication devices, voice recognition conference systems, event venue guidance broadcast devices, television receivers, radio receivers, public facility guidance broadcast facilities It can be easily applied to broadcasting facilities for elderly and handicapped facilities, school broadcasting facilities, in-car broadcasting of trains and buses, and in-house broadcasting in places where many people gather, such as stations, shopping centers, department stores, movie theaters, etc. Excellent speech clarification effect can be obtained.

  According to the voice processing device 100 and the voice clarification device 200 of the present invention, not only the voice that is difficult to hear is clarified, but also the voice color unique to the individual of the sender can be maintained, so that natural voice without a sense of incongruity can be maintained. Can be generated.

DESCRIPTION OF SYMBOLS 100 ... Audio processing apparatus 200 ... Audio | voice clarifying apparatus 1 ... Microphone 2 ... Input buffer 3 ... Phase corrector 4 ... High pass filter 5 ... Low pass filter 6 ... Effect adjuster 7 ... Adder / synthesizer 8 ... Output buffer 9 ... Attenuation 10: Audio output terminal 11 ... Audio processing unit 12 ... Power amplifier 13 ... Speaker 15 ... Input terminal 16 ... Input ground terminal 17 ... Output terminal 18 ... Output ground terminal 19 ... Operational amplifier 20 ... Modulator equipped with the device of the present invention Unit 21: Wireless transmission unit 22 ... Transmitting antenna or infrared light emitting unit 23 ... Reception antenna or infrared light receiving unit 24 ... Radio signal receiving unit 30 ... Bandpass filter 31 ... Input adder 32 ... Mixing buffer 33 ... Housing 34 ... Input side connection port 35 ... Output side connection port

Claims (6)

A bandpass filter that branches the collected audio signal and removes a low-frequency component below a low-order formant and a high-frequency component exceeding a high-order formant from one of the branched audio signals;
A phase corrector that corrects the phase of the other branched audio signal so as to match the phase of the audio signal that has passed through the bandpass filter;
An audio processing apparatus comprising: an adder / synthesizer that adds and combines the audio signal that has passed through the bandpass filter with the audio signal that has passed through the phase corrector. A bandpass filter that removes low-frequency components below the low-order formant and high-frequency components exceeding the high-order formant from the left and right audio signals to be collected stereo signals;
A pair of phase correctors that respectively correct the phase of the left and right audio signals to match the phase of the audio signal that has passed through the bandpass filter;
A speech processing apparatus, comprising: a pair of adder / synthesizers that respectively add and synthesize an audio signal that has passed through the bandpass filter and a left / right audio signal that has passed through each phase corrector. The speech processing apparatus according to claim 1 or 2,
The bandpass filter is a high-pass filter having a function of removing a low-frequency component below a low-order formant from one branched audio signal and amplifying the signal,
An audio processing apparatus comprising: a low-pass filter that removes a high-frequency component exceeding a high-order formant from an audio signal that has passed through the high-pass filter. The speech processing apparatus according to any one of claims 1 to 3,
An audio processing apparatus comprising an effect adjuster for adjusting a value of an audio signal before synthesis that has passed through the bandpass filter.   An input side connection port for accommodating the sound processing device according to any one of claims 1 to 4 in a portable housing, and detachably connecting a sound collecting microphone to the surface of the housing. And an audio clarification device comprising: an output side connection port that outputs an audio signal processed by the audio processing device to broadcasting equipment. A first step of branching the collected audio signal;
A second step of removing low-frequency components below the low-order formant and high-frequency components exceeding the high-order formant from the one audio signal branched in the first step;
A third step of correcting the phase of the other audio signal branched in the first step so as to match the phase of the audio signal processed in the second step;
A voice processing method comprising: a fourth step of synthesizing and outputting the voice signal processed in the second step and the voice signal processed in the third step.

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