WO2013084810A1 - 耳孔装着型収音装置、信号処理装置、収音方法 - Google Patents

耳孔装着型収音装置、信号処理装置、収音方法 Download PDF

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Publication number
WO2013084810A1
WO2013084810A1 PCT/JP2012/081054 JP2012081054W WO2013084810A1 WO 2013084810 A1 WO2013084810 A1 WO 2013084810A1 JP 2012081054 W JP2012081054 W JP 2012081054W WO 2013084810 A1 WO2013084810 A1 WO 2013084810A1
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WO
WIPO (PCT)
Prior art keywords
sound
signal
microphone
collected
ear
Prior art date
Legal status (The legal status is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the status listed.)
Ceased
Application number
PCT/JP2012/081054
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English (en)
French (fr)
Japanese (ja)
Inventor
宏平 浅田
慎平 土谷
大善 小林
耕治 投野
Current Assignee (The listed assignees may be inaccurate. Google has not performed a legal analysis and makes no representation or warranty as to the accuracy of the list.)
Sony Corp
Original Assignee
Sony Corp
Priority date (The priority date is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the date listed.)
Filing date
Publication date
Application filed by Sony Corp filed Critical Sony Corp
Priority to EP12854783.3A priority Critical patent/EP2790416B1/en
Priority to US14/360,948 priority patent/US9237392B2/en
Priority to CN201280058984.8A priority patent/CN103959813B/zh
Publication of WO2013084810A1 publication Critical patent/WO2013084810A1/ja
Anticipated expiration legal-status Critical
Priority to US14/992,906 priority patent/US9918162B2/en
Priority to US15/883,667 priority patent/US11070910B2/en
Priority to US17/303,673 priority patent/US11765497B2/en
Priority to US18/446,275 priority patent/US12501208B2/en
Ceased legal-status Critical Current

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    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R1/00—Details of transducers, loudspeakers or microphones
    • H04R1/20—Arrangements for obtaining desired frequency or directional characteristics
    • H04R1/32—Arrangements for obtaining desired frequency or directional characteristics for obtaining desired directional characteristic only
    • H04R1/40—Arrangements for obtaining desired frequency or directional characteristics for obtaining desired directional characteristic only by combining a number of identical transducers
    • H04R1/406—Arrangements for obtaining desired frequency or directional characteristics for obtaining desired directional characteristic only by combining a number of identical transducers microphones
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L21/00—Speech or voice signal processing techniques to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility
    • G10L21/02—Speech enhancement, e.g. noise reduction or echo cancellation
    • G10L21/0208—Noise filtering
    • G10L21/0216—Noise filtering characterised by the method used for estimating noise
    • G10L21/0232—Processing in the frequency domain
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L25/00—Speech or voice analysis techniques not restricted to a single one of groups G10L15/00 - G10L21/00
    • G10L25/78—Detection of presence or absence of voice signals
    • G10L25/84—Detection of presence or absence of voice signals for discriminating voice from noise
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R1/00—Details of transducers, loudspeakers or microphones
    • H04R1/10—Earpieces; Attachments therefor ; Earphones; Monophonic headphones
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R1/00—Details of transducers, loudspeakers or microphones
    • H04R1/10—Earpieces; Attachments therefor ; Earphones; Monophonic headphones
    • H04R1/1016—Earpieces of the intra-aural type
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R1/00—Details of transducers, loudspeakers or microphones
    • H04R1/10—Earpieces; Attachments therefor ; Earphones; Monophonic headphones
    • H04R1/1058—Manufacture or assembly
    • H04R1/1075—Mountings of transducers in earphones or headphones
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R3/00—Circuits for transducers, loudspeakers or microphones
    • H04R3/005—Circuits for transducers, loudspeakers or microphones for combining the signals of two or more microphones
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R3/00—Circuits for transducers, loudspeakers or microphones
    • H04R3/04—Circuits for transducers, loudspeakers or microphones for correcting frequency response
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R11/00—Transducers of moving-armature or moving-core type
    • H04R11/02—Loudspeakers
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R2201/00—Details of transducers, loudspeakers or microphones covered by H04R1/00 but not provided for in any of its subgroups
    • H04R2201/003—Mems transducers or their use
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R2225/00—Details of deaf aids covered by H04R25/00, not provided for in any of its subgroups
    • H04R2225/43—Signal processing in hearing aids to enhance the speech intelligibility
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R2410/00—Microphones
    • H04R2410/03—Reduction of intrinsic noise in microphones
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R2430/00—Signal processing covered by H04R, not provided for in its groups
    • H04R2430/20—Processing of the output signals of the acoustic transducers of an array for obtaining a desired directivity characteristic
    • H04R2430/23—Direction finding using a sum-delay beam-former
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R25/00—Deaf-aid sets, i.e. electro-acoustic or electro-mechanical hearing aids; Electric tinnitus maskers providing an auditory perception
    • H04R25/40—Arrangements for obtaining a desired directivity characteristic
    • H04R25/407—Circuits for combining signals of a plurality of transducers

Definitions

  • This technique is a signal for performing signal processing on a sound collecting device having an ear hole mounting type sound collecting device having a mounting part configured to be at least partially insertable into the ear hole part, and an internal microphone provided in the mounting part.
  • the present invention relates to a processing apparatus and a sound collection method.
  • an earphone microphone (earphone with a built-in microphone) for enabling listening to received sound and collecting sound is used.
  • FIG. 16 shows an example of a general earphone microphone (hereinafter referred to as a conventional earphone microphone 100) that is widely used at present.
  • the conventional earphone microphone 100 is provided with an earphone unit 101 for listening to the received sound and a microphone 102A for collecting the uttered sound.
  • Earphone unit 101 is configured to be worn on the ear of wearer H, and has a built-in speaker for outputting a received sound.
  • a cord upper housing 102 is formed on a cord for transmitting a signal to the earphone unit 101, and a microphone 102 ⁇ / b> A is formed in the cord upper housing 102.
  • the uttered sound emitted from the wearer (speaker) reaches the microphone 102A via the outside (outside air) and is collected.
  • the microphone 102A for collecting the uttered sound is exposed to the outside. That is, the microphone 102A is in direct contact with external noise (environmental noise). For this reason, in the conventional earphone microphone 100, ambient noise is picked up relatively loudly together with the uttered sound, and the S / N (signal-to-noise ratio) of the utterance signal tends to decrease. That is, as a result, it is difficult for the other party of the call to hear the speech from the wearer H.
  • noise reduction processing such as SS method (SS: Spectrum Subtraction) on the uttered sound collection signal.
  • SS method Spectrum Subtraction
  • noise reduction processing relatively large processing resources are required, which is disadvantageous in terms of product cost and power consumption.
  • noise reduction processing involving nonlinear processing on the frequency axis as in the SS method generally has a problem of deterioration in sound quality after processing.
  • the present technology has been made in view of the above-mentioned problems, and the problem is to realize a sound pickup with a good S / N with reduced influence of noise, without using noise reduction processing.
  • the present technology is configured as follows as an ear hole mounting type sound pickup device. That is, at least a part of the mounting part is configured to be insertable into the ear canal part, and the mounting part is configured so that a substantially sealed internal space connected to the external auditory canal is formed inside the ear hole part in a mounted state. Is provided. In addition, an internal microphone is provided that is disposed in the internal space of the mounting portion and picks up the uttered sound that is emitted by the wearer and propagates through the ear canal in a state of being attached to the ear canal portion.
  • a low-frequency extraction filter unit that performs a filtering process for extracting a low-frequency component from the collected sound signal by the internal microphone, or a high-frequency emphasizing equalizing process for the sound-collected signal by the internal microphone.
  • One of the equalizing processing sections to be applied is provided.
  • the microphone that collects the uttered sound (the internal microphone) is substantially sealed from the outside, and is installed in a space that communicates with the ear canal of the wearer (speaker). By being installed in a space sealed from the outside, the influence of noise can be effectively reduced. Further, since the utterance sound propagating through the ear canal of the wearer is collected, the conventional earphone microphone (FIG. 16) that collects the utterance sound emitted from the wearer and propagating in the outside world is collected. Can pick up the utterance sound with good S / N. In addition, in the present technology, the low-frequency component of the sound-collected signal from the internal microphone is extracted by the low-frequency extraction filter unit.
  • the equalizing processing unit is provided. By providing the equalizing processing unit, it is possible to reduce the humming sound that occurs when the uttered sound via the ear canal is collected and to improve the sound quality of the uttered sound collected signal.
  • an utterance sound can be collected with better S / N than a conventional earphone microphone that collects an utterance sound propagating in the outside world. Further, according to the present technology, noise reduction processing for a collected sound signal can be made unnecessary, and as a result, increase in signal processing resources can be prevented, which is advantageous in terms of product cost and power consumption.
  • FIG. 1 It is a figure for demonstrating the structure of the mounting part with which the sound collection system of embodiment is provided. It is the figure which showed typically the mode of the sound collection of the speech sound by the sound collection system of embodiment. It is a figure for demonstrating the structure of the signal processing system for sound quality improvement. It is a figure for demonstrating the specific frequency characteristic which should be set to an equalizer for sound quality improvement. It is explanatory drawing about a compressor process. It is a figure for demonstrating the point that the speech sound component becomes more dominant than an external noise component in the low region of the sound collection signal by an internal microphone. It is the figure which showed the structure of the sound collection system as Example 1.
  • FIG. 1 It is a figure for demonstrating the structure of the mounting part with which the sound collection system of embodiment is provided. It is the figure which showed typically the mode of the sound collection of the speech sound by the sound collection system of embodiment. It is a figure for demonstrating the structure of the signal processing system for sound quality improvement. It is a figure for demonstrating the specific frequency characteristic which should be set
  • FIG. 10 is a flowchart illustrating a specific processing procedure to be executed by a control unit according to the fifth embodiment. It is the figure which showed the structure of the sound collection system as Example 6. FIG. It is the figure which showed the structural example of the conventional earphone microphone.
  • FIG. 1 is a diagram for describing a structure of a mounting portion 1 included in a sound collection system as an embodiment according to the present technology. Specifically, A in FIG. 1 represents a perspective view of the wearing portion 1, and B in FIG. 1 represents the ear canal HA and ear canal HB of the wearer H in the wearing state of the wearer (speaker) H. The relationship with the mounting part 1 is represented by a sectional view.
  • the wearing unit 1 is provided with an internal microphone 1 ⁇ / b> B in order to pick up the utterance sound of the wearer (speaker) H.
  • a MEMS microphone MEMS: Micro Electro Mechanical Systems
  • the internal microphone 1B is adopted as the internal microphone 1B in consideration of the arrangement space.
  • the outer shape of the mounting portion 1 is configured such that at least a part of the mounting portion 1 can be inserted into the ear hole portion of the wearer H so that the mounting portion 1 can be worn on the ear portion of the wearer H.
  • the mounting part 1 in this case is formed with an ear hole insertion part 1A that is shaped to be inserted into the ear hole part HB of the wearer H, and the ear hole insertion part 1A corresponds to the ear hole part HB.
  • the wearing part 1 is put into a wearing state with respect to the ear part of the wearer H.
  • the mounting part 1 is comprised so that the internal space 1V connected with the external ear canal HA of the wearer H may be formed as shown to B of FIG.
  • the ear hole insertion portion 1A of the mounting portion 1 is covered with an elastic material on the surface portion, similarly to the ear hole insertion portion of the canal type earphone portion, and is in close contact with the ear hole portion HB at the time of mounting. It is comprised so that can be obtained.
  • the internal space 1V is a substantially sealed space from the outside.
  • the internal microphone 1B is arranged in the internal space 1V.
  • FIG. 2 is a diagram schematically showing a state of sound collection by the sound collection system according to the embodiment having the mounting portion 1.
  • the noise that propagates from the outside of the housing can be prevented. Since the sound insulation property of the microphone is sufficiently enhanced, the mixing of noise into the internal microphone 1B is sufficiently suppressed. That is, as a result, the utterance sound can be collected with a better S / N (signal-to-noise ratio) than the conventional earphone microphone 100 (see FIG. 13) that collects the utterance sound via the outside world. Note that the sound insulation at this time is not limited as long as it can cover at least the band of noise intended to be suppressed.
  • the gain (response) in the low frequency is larger in the external auditory canal HA than in the normal free space.
  • the response characteristic of the low range becomes relatively large. Due to this influence, the transmitted voice based on the collected sound signal from the internal microphone 1B becomes a sound that is confined to the low frequency range, and it becomes somewhat difficult for the other party of the call to hear.
  • signal processing means as an equalizer (EQ) as shown in FIG.
  • EQ equalizer
  • FIG. 4 is a diagram for explaining specific frequency characteristics to be set in the equalizer 11.
  • a specified conversation sequence is collected by a microphone installed in the outside of the wearing unit 1 in a no-noise state.
  • the frequency characteristics of the collected sound signal (a set of ⁇ plots and broken lines in the figure) and the same conversation sequence are collected by the internal microphone 1B in the internal space 1V connected to the ear canal HA in the same noise-free state.
  • the frequency characteristics of the collected sound signal at the time of the sound ( ⁇ plot and a set of alternate long and short dash lines) are shown in comparison. In this figure, the frequency characteristic shows a time-averaged value on the frequency axis.
  • the diaphragm of the internal microphone 1B is compared with the external environment as an unsealed environment.
  • the amplitude will be large.
  • a larger microphone output voltage can be obtained in the lower range than a microphone installed outside.
  • the sound pickup signal ( ⁇ & alternate long and short dash line) by the internal microphone 1B has a low-frequency bulge compared to the sound pickup signal ( ⁇ & broken line) by the microphone installed outside. Can be confirmed. If the collected sound signal of the internal microphone 1B having the characteristics shown in FIG. 4A is used as it is, the transmitted sound to the other party of the call will be a voice with low intelligibility as a muffled sound. There is a risk that listening at will be difficult.
  • the clarity of the transmitted sound to be heard by the other party is improved.
  • the frequency characteristic of the sound collected signal by the internal microphone 1B should be close to the frequency characteristic of the sound collected signal by the microphone installed in the outside world.
  • a filter that is, an equalizer 11 represented by a transfer function as shown in B of FIG. 4 is prepared, and the frequency characteristic of the collected sound signal of the internal microphone 1B is corrected by the filter. That is, the sound pickup signal frequency characteristic of the internal microphone 1B may be corrected by an equalizer 11 having a high frequency emphasis type (low frequency suppression type) filter characteristic as shown in FIG. Accordingly, it is possible to obtain natural speech with higher clarity after the equalizer than before the equalizer.
  • FIG. 4A the frequency characteristics of the collected sound signal of the internal microphone 1B after correction by the equalizer 11 having the filter characteristics shown in FIG. Yes.
  • the frequency characteristics it can be seen that the sound collection signal from the internal microphone 1B approaches the sound collection signal from the microphone installed in the external environment, and has a more natural frequency characteristic balance.
  • the noise collection processing by the noise gate processing unit 12 is performed on the collected sound signal by the internal microphone 1 ⁇ / b> B via the microphone amplifier 10, and then the characteristic correction by the equalizer 11 is performed. ing. In addition, the compressor process by the compressor 13 is performed on the collected sound signal via the equalizer 11.
  • the noise gate processing unit 12 lowers the output signal level (that is, closes the gate) when the level of the input signal falls below a certain level, and restores the output signal level when the level exceeds the certain level (gate). Open).
  • parameters such as the rate of attenuation of the output level in noise gate processing, the opening / closing envelope of the gate, and the frequency band to which the gate reacts are appropriately set so as to improve the clarity of the uttered sound. Set.
  • the compressor 13 performs processing for adjusting the time axis amplitude of the input sound pickup signal as the compressor processing.
  • FIG. 5A the time axis waveform of the collected sound signal before the compressor process is shown by A in FIG. 5, and the time axis waveform of the collected sound signal after the compressor process is shown by B in FIG.
  • the equalizer 11 described above is intended to improve the sound quality by adjusting the frequency characteristics of the collected sound signal, but the compressor process is to correct the waveform of the collected sound signal on the time axis.
  • the uttered voice reaches the diaphragm of the internal microphone 1B through the external auditory canal HA through vibration in the human body such as the bone and meat of the wearer H. This is to some extent compared with air propagation. It will have non-linearity. For this reason, the difference in the size of the utterance voice that changes depending on the volume of the voice at the time of utterance is larger than when collecting sound via normal air propagation, and if it is left as it is, it is difficult to hear the collected sound. There is a risk of becoming. Referring to FIG. 5A, it can be confirmed that the difference in the size of the speech between the speech groups to be spoken is large.
  • the compressor 13 adjusts the time axis amplitude of the collected sound signal from the internal microphone 1B as shown in FIG. That is, the difference in the size of the uttered voice is suppressed. This makes it easier to listen to the uttered voice and improves the sound quality.
  • various signal processing for the collected sound signal may be realized by an analog electric circuit, or may be realized by digital signal processing via an ADC (A / D converter).
  • FIG. 6 is an explanatory diagram regarding this point.
  • a frequency characteristic of a sound pickup signal by the internal microphone 1B a frequency characteristic in a non-speech portion under a general noise environment (a set of ⁇ and a broken line: noise only), a speech portion The frequency characteristics (a set of ⁇ and a solid line: noise + speech sound) are shown.
  • general airplane cabin noise was used as the noise.
  • the unit of analysis is 1/3 octave.
  • a signal when noise and a speech sound are collected ( ⁇ & solid line) rather than a signal when only noise is collected ( ⁇ & broken line). It can be confirmed that the level is higher especially in the low range. That is, when the utterance sound is collected via the external auditory canal HA by the internal microphone 1B, the utterance sound is more dominant than the external noise particularly in the low frequency range of the collected sound signal (in the figure, the internal microphone 1). Voice dominant band). This is because the noise component is attenuated particularly in the low range by the sealing / sound insulation function derived from the structure of the mounting portion 1, while the sound collection component via the ear canal HA as shown in FIG. Depends on the gain of the low range becoming large.
  • the collected sound signal by the internal microphone 1B is subjected to the filtering process, and the low frequency component of the collected sound signal (the component of the voice dominant band of the internal microphone 1B) is extracted. Further S / N improvement can be achieved.
  • FIG. 7 is a diagram showing an example of the configuration of a sound collection system (hereinafter referred to as Example 1) as an embodiment for further improving the S / N by using the low-frequency component filtering process as described above. is there.
  • Example 1 a sound collection system
  • the sound collection system as the first embodiment includes a mounting unit 1 and a signal processing unit 2.
  • a speaker 1S for outputting a received sound is disposed in the internal space 1V of the mounting unit 1 together with the internal microphone 1B.
  • the speaker 1S is a BA (balanced armature) type considering the installation space.
  • the signal processing unit 2 includes the above-described microphone amplifier 10, equalizer 11, noise gate processing unit 12, and compressor 13, and also includes an LPF (low-pass filter) 14 and an amplifier 15.
  • the LPF 14 is disposed between the microphone amplifier 10 and the noise gate processing unit 12, and thereby performs low-pass filtering processing on the collected sound signal by the internal microphone 1 ⁇ / b> B after passing through the microphone amplifier 10.
  • the cut-off frequency of the LPF 14 is appropriately set so that a component of “sound dominant band of the internal microphone” as shown in FIG. 5 can be extracted.
  • the sound pickup signal by the internal microphone 1 ⁇ / b> B after passing through the compressor 13 is output to the outside of the signal processing unit 2 as a transmission signal as shown in the figure.
  • the signal processing unit 2 is supplied with an incoming signal from the outside.
  • the amplifier 15 amplifies the received signal and drives the speaker 1S in the mounting unit 1 based on the amplified received signal. As a result, a reception sound corresponding to the reception signal is output from the speaker 1S.
  • the S / N of the utterance sound collection signal is ensured by the (passive) sound insulation performance of the housing of the mounting unit 1 against the environmental noise, and then the internal microphone.
  • the S / N of the utterance sound collection signal is ensured by the (passive) sound insulation performance of the housing of the mounting unit 1 against the environmental noise, and then the internal microphone.
  • Example 1 the effect that it becomes easy for the wearer H to hear a received sound by the sound insulation effect which the mounting part 1 has is also acquired.
  • the signal processing unit 2 for realizing the filtering process for extracting the voice dominant band component and the various signal processes (equalizer 11 to compressor 13) for improving the sound quality as described above.
  • the various signal processes for improving the sound quality as described above.
  • FIG. 8 is a diagram showing a configuration example of each of “integrated type” and “separated type”.
  • the signal processing unit 2 is provided inside the housing of the mounting unit 1.
  • a transmission signal that is, a sound collection signal by the internal microphone 1B after various signal processing by the signal processing unit 2
  • the external device 50 for example, an information processing device such as a smartphone.
  • a reception signal is transmitted from the external device 50 to the mounting unit 1.
  • the signal processing unit 2 is provided inside the external device 50.
  • a sound collection signal (transmission sound collection signal in the figure) from the internal microphone 1 is transmitted from the mounting unit 1 to the external device 50.
  • the external device 50 transmits the reception signal (the reception sound output signal in the figure) after amplification by the amplifier 15 in the signal processing unit 2 to the mounting unit 1 (speaker 1S).
  • FIG. 9 is an explanatory diagram of a configuration of a sound collection system according to the second embodiment.
  • the S / N of the utterance / acquisition signal is further improved by the beam forming process using the signals collected by the left and right channels, and the received sound is heard in both ears of the wearer H. It is to make.
  • the channel is also referred to as ch.
  • the received signal is generally monaural. Therefore, in the second embodiment, a system for listening to the monaural reception sound with both ears is proposed.
  • the sound collection system according to the second embodiment includes a mounting unit 3 and a signal processing unit 20 instead of the signal processing unit 2 as compared with the sound collection system according to the first embodiment illustrated in FIG.
  • the point is different.
  • the mounting portion 3 is to be mounted on the ear of the wearer H that is opposite to the ear on the side on which the mounting portion 1 is mounted. As with the mounting unit 1, at least a part of the mounting unit 3 can be inserted into the ear hole HB of the wearer H and can be mounted on the ear of the wearer H.
  • the mounting part 3 is also formed with an ear hole insertion part 3A that can be inserted into the ear hole part HB of the wearer H, and the ear hole insertion part 3A is inserted into the ear hole part HB.
  • the wearing unit 3 is put into a wearing state with respect to the ear part of the wearer H.
  • the wearing portion 3 is also configured to form an internal space 3V connected to the ear canal HA of the wearer H in the wearing state to the wearer H, and the ear hole insertion portion 3A has a surface portion thereof. It is covered with an elastic material so that it can be in close contact with the ear canal HB when worn.
  • An internal microphone 3B is arranged in the internal space 3V of the mounting portion 3 as shown in the figure. In the case of this example, a MEMS microphone is also used for the internal microphone 3B.
  • a speaker 3 ⁇ / b> S is disposed in the internal space 3 ⁇ / b> V of the mounting portion 3.
  • the speaker S3 is also of the BA type.
  • the speaker 3 ⁇ / b> S is driven based on the received signal amplified by the amplifier 15 provided in the signal processing unit 20.
  • the output of the amplifier 15 is also supplied to the speaker 1S on the mounting unit 1 side in the same manner as in the first embodiment.
  • the received sound based on the received signal is transmitted to the mounting unit 1 side and the mounting unit 3. Will be output from both sides.
  • Example 2 the mounting portion 1 side is assumed to be the Lch side, and the mounting portion 2 side is assumed to be the Rch side.
  • the signal processing unit 20 is different from the signal processing unit 2 in the first embodiment in that an Rch microphone amplifier 21 and an LPF 22 and a beam forming processing unit 23 are added.
  • the microphone amplifier 21 amplifies a sound collection signal from the internal microphone 3B on the mounting unit 3 side.
  • the LPF 22 performs a low-pass filtering process for extracting a low-frequency component as the above-described audio dominant band with respect to a sound-collected signal from the internal microphone 3B with a cutoff frequency equivalent to that of the LPF 14. In this case, the LPF 22 performs a low-pass filtering process on the collected sound signal by the internal microphone 3B after being amplified by the microphone amplifier 21.
  • the LPF 22 can also improve the S / N of the sound collected signal from the internal microphone 3B.
  • the beamforming processing unit 23 collects a sound collection signal (Lch side sound collection signal) from the internal microphone 1B that has passed through the LPF 14 disposed on the Lch side, and a sound collection signal (Rch) from the internal microphone 3B that has passed through the LPF 22 disposed on the Rch side. Side sound pickup signal), and beam forming processing is performed.
  • a sound collection signal Lch side sound collection signal
  • Rch sound collection signal
  • the process of adding the Lch side sound collection signal and the Rch side sound collection signal can be most simply mentioned.
  • the internal microphone 1B that performs speech collection on the Lch side and the internal microphone 3B that performs speech collection on the Rch side are approximately equidistant from the mouth (voice zone) of the wearer H that is the speech source. Therefore, the beam forming processing unit 23 adds the collected sound signals to efficiently extract the sound from the direction of the utterance sound source (via the external auditory canal HA) and from other directions. Sound (noise component) can be suppressed. That is, further S / N improvement of the utterance sound pickup signal can be achieved.
  • a sound component from the sound source direction is determined from the sound analysis result of the collected sound signal, and only the sound component from the sound source direction is determined from the determination result.
  • An extraction method can be used.
  • a process of determining a dominant component in the collected sound signal may be performed.
  • the collected sound signal after the beamforming processing by the beamforming processing unit 23 is output to the outside of the signal processing unit 20 as an utterance signal via the noise gate processing unit 12 ⁇ the equalizer 11 ⁇ the compressor 13.
  • the improvement effect due to the (passive) sound insulation performance of the housings of the mounting portions 1 and 3 and the LPFs 14 and 22 In addition to the improvement effect obtained by extracting the dominant band component of the uttered voice, the S / N improvement effect obtained by the reduction of the noise component by the beamforming processing unit 23 can be obtained.
  • Example 1 Since the sound-insulation effect by the mounting part 3 is also acquired, the sound-insulation effect can be acquired in both ears of the wearer H. As a result, the effect of Example 1 is obtained. Compared with the case, the ease of listening to the received sound is improved.
  • the signal processing for further improving the S / N of the uttered sound collection signal includes, for example, the noise reduction processing by the SS method (SS: Spectrum Subtraction) in addition to the beam forming processing described above. It is good also as what performs.
  • the noise reduction processing of the SS method is disclosed in, for example, Reference Document 1 below. ⁇ Reference 1 ... JP 2010-11117 A
  • the second embodiment can adopt both “integrated” and “separated” configurations as shown in FIG.
  • the signal processing unit 20 is one of the mounting portions 1 and 3.
  • a sound collection signal from an internal microphone in the other mounting portion is input to one mounting portion where the signal processing unit 20 is provided, and the amplifier 15 is connected from the one mounting portion to the other mounting portion. The received signal after amplification by is input.
  • the configuration in which the monaural transmission signal is obtained by performing the beam forming process as in the second embodiment only the configuration (23, 12, 11, 13) below the beam forming processing unit 23 is used for any of the mounting units 1, 3.
  • it may be provided (in other words, only the microphone amplifier 21 and the LPF 22 are provided in the mounting portion 3 among the components constituting the signal processing unit).
  • FIG. 10 is a diagram illustrating a configuration of a sound collection system according to the third embodiment.
  • an external microphone 1C is added to the mounting unit 1 and a signal processing unit 25 is provided instead of the signal processing unit 2.
  • the external microphone 1 ⁇ / b> C is a microphone installed so as to be able to pick up sound generated outside the housing of the mounting unit 1.
  • the external microphone 1 ⁇ / b> C is installed so that the sound collection port is located on the housing surface of the mounting unit 1.
  • the MEMS microphone is used for the external microphone 1C as well as the internal microphone 1B.
  • the external microphone 1 ⁇ / b> C only needs to be installed so as to be able to collect sound generated outside the housing of the mounting unit 1, and the sound collecting port is not necessarily directly exposed to the outside of the housing of the mounting unit 1. There is no need.
  • the signal processing unit 25 is different from the signal processing unit 2 in that a microphone amplifier 26, an HPF (high pass filter) 27, a delay circuit (DELAY in the figure) 28, and an addition unit 29 are added.
  • the microphone amplifier 26 amplifies a sound collection signal from the external microphone 1C.
  • the HPF 27 performs a high-pass filtering process on the collected sound signal from the external microphone 1 ⁇ / b> C after being amplified by the microphone amplifier 26.
  • the delay circuit 28 is provided in a signal processing system (between the microphone amplifier 10 and the adder 29) for a sound collected signal by the internal microphone 1B, and delays the sound collected signal by the internal microphone 1B by a predetermined time length.
  • the delay circuit 28 is provided between the LPF 14 and the adder 29 and is configured to give a delay of a predetermined time length to the collected sound signal by the internal microphone 1B after passing through the LPF 14.
  • the addition unit 29 is provided to add the sound collection signal from the internal microphone 1B that has undergone the low-pass filtering process by the LPF 14 and the sound collection signal from the external microphone 1C that has undergone the high-pass filtering process by the HPF 27.
  • the adder 29 in this case is provided at a position where the output signal of the delay circuit 28 and the output signal of the HPF 27 are added.
  • the addition signal by the addition unit 29 is output to the outside of the signal processing unit 25 as an utterance signal after passing through the noise gate processing unit 12 and the compressor 13.
  • the equalizer 11 that is, the equalization filter for suppressing the low-frequency swell (bumping sound) caused by the sound collection via the external ear canal HA by the internal microphone 1B is provided for the sound collection signal side by the internal microphone 1B. And is arranged on the front side of the adder 29 (that is, before synthesis with the output of the HPF 27). Specifically, in the present example, the equalizer 11 is arranged between the microphone amplifier 10 and the LPF 14 and performs an equalization process on the collected sound signal from the internal microphone 1B after being amplified by the microphone amplifier 10. .
  • the external microphone 1C is provided in the mounting unit 1 and the signal obtained by performing high-pass filtering processing by the HPF 27 on the collected sound signal is added to the LPF 14 by the adding unit 29. It is assumed that it is added to the collected sound signal from the internal microphone 1B.
  • the uttered sound emitted from the mouth of the wearer H is collected through the outside world (outside air).
  • the external microphone 1C also picks up environmental noise.
  • the high-pass filtering processing by the HPF 27 is applied to the sound collection signal from the external microphone 1C.
  • the sound collection signal from the external microphone 1C is, in contrast to the sound collection signal from the internal microphone 1B, its middle / high range ( This is because the speech component is superior to the noise component in the middle range and the high range.
  • FIG. 11 is a diagram for explaining this point.
  • the frequency characteristics in the non-speech part under the general noise environment ( ⁇ and broken lines) are shown as the frequency characteristics of the collected sound signal by the external microphone 1C. Set: noise only) and frequency characteristics in the utterance part ( ⁇ and solid line set: noise + utterance sound).
  • FIG. 11B as a comparison, the frequency characteristic in the non-speech portion under the general noise environment for the collected sound signal by the internal microphone 1B, similar to that shown in FIG. : Only noise), and frequency characteristics in the utterance part (a pair of ⁇ and solid line: noise + utterance sound).
  • the actual noise (• & broken line) such as the noise in the airplane has a very low frequency component, and the level tends to decrease as the frequency goes up. For this reason, in sound collection by the external microphone 1C, the uttered voice component tends to be superior to the noise component in a relatively middle / high range.
  • the high-pass filtering process to the collected sound signal of the external microphone 1C by the configuration as in the third embodiment, it is possible to obtain a middle / high frequency utterance sound of the wearer H.
  • a component can be extracted with a relatively good S / N ratio.
  • the sound collection signal that has passed through the HPF 27 and the sound collection signal that has passed through the LPF 14 are added by the adder 29. That is, for each of the external and internal sound collection microphone output signals, a band superior to the sound collection is selected and added.
  • the configuration of the third embodiment as described above it is possible to add not only the low range of the utterance sound but also the effective information of the middle / high range as the utterance collection signal. It is possible to listen to speech sound with higher sound quality.
  • the cut-off frequency of the HPF 27 is appropriately set so as to extract the middle / high-frequency speech dominant band components as shown in FIG.
  • the delay circuit 28 is provided to delay the sound collection signal from the internal microphone 1B with respect to the sound collection signal from the external microphone 1C. This is because the internal microphone 1B and the external microphone 1C are installed. This is intended to absorb the difference in the arrival time of the uttered voice accompanying the difference in position. That is, the delay circuit 28 is set with a delay time corresponding to the time difference between the arrival time of the utterance sound of the wearer H to the internal microphone 1B and the arrival time to the external microphone 1C. It is intended to suppress deterioration in sound quality that may occur when the distance between 1B and the external microphone 1C is relatively large and the difference in arrival time is relatively long. For example, when the distance between the microphones is 1 cm, if the sound speed is about 340 m / sec, a delay time of about 30 ⁇ sec should be set.
  • FIG. 12 is a diagram illustrating a configuration of a sound collection system according to the fourth embodiment.
  • the processing characteristics of various signal processing units for improving S / N and sound quality are made variable, and these processing characteristics can be switched as necessary. Appropriate improvement processing according to the external noise situation and the intention of the user (wearer H) is realized.
  • the processing characteristics of each unit are switched according to a user operation.
  • the sound collection system in this case is different from the sound collection system of the third embodiment (FIG. 10) in that a signal processing unit 30 is provided instead of the signal processing unit 25. In addition, a memory 32 is newly added.
  • the signal processing unit 30 is different from the signal processing unit 25 in that the processing characteristics of the equalizer 11, the LPF 14, the HPF 27, the noise gate processing unit 12, and the compressor 13 are variable.
  • each of the above-described parts whose processing characteristics are variable is represented as an equalizer 11 ', LPF 14', HPF 27 ', noise gate processing unit 12', and compressor 13 'as shown in the figure.
  • the signal processing unit 30 is provided with a control unit 31.
  • the control unit 31 performs switching control of processing characteristics of the equalizer 11 ′, the LPF 14 ′, the HPF 27 ′, the noise gate processing unit 12 ′, and the compressor 13 ′.
  • a mode instruction signal is input from the outside to the control unit 31 in this case.
  • This mode instruction signal is a signal indicating another processing mode selected in accordance with the user operation.
  • the memory 32 is a storage device that can be read by the control unit 31.
  • mode-processing characteristic correspondence information 32A in the figure information on each mode that can be instructed by the mode instruction signal and the mode are provided.
  • Information relating to the processing characteristics to be set in the respective sections (equalizer 11 ′, LPF 14 ′, HPF 27 ′, noise gate processing section 12 ′, and compressor 13 ′) whose processing characteristics are variable (hereinafter referred to as processing characteristics information and Information) is stored.
  • processing characteristic information for example, parameter information necessary for changing the processing characteristics of the above-described units may be stored.
  • the control unit 31 reads the processing characteristic information corresponding to the characteristic indicated by the mode instruction signal, and changes the processing characteristic of each part in which the processing characteristic is variable according to the processing characteristic information.
  • S / N / sound quality improvement processing can be performed in an appropriate processing mode intended by the user according to an external noise situation or the like.
  • FIG. 13 is a diagram illustrating a configuration of a sound collection system according to the fifth embodiment.
  • the processing characteristics are automatically switched regardless of the user operation based on the result of performing the voice analysis on the external noise situation.
  • the sound collection system according to the fifth embodiment is different from the sound collection system according to the fourth embodiment in that a signal processing unit 35 is provided instead of the signal processing unit 30. Further, the analysis result-processing characteristic correspondence information 32B is stored in the memory 32 instead of the mode-processing characteristic correspondence information 32A.
  • the signal processing unit 35 is different from the signal processing unit 30 of the fourth embodiment in that a control unit 36 is provided instead of the control unit 31.
  • the control unit 36 performs a voice analysis process on the external noise based on the collected sound signal from the external microphone 1C, and based on the analysis result and the information content of the analysis result-processing characteristic correspondence information 32B, the equalizer 11 ′, the LPF 14 ′, The processing characteristics of the HPF 27 ′, the noise gate processing unit 12 ′, and the compressor 13 ′ are switched. As shown in the figure, in the case of this example, the sound collection signal from the external microphone 1 ⁇ / b> C before input is input to the microphone amplifier 26 to the control unit 36.
  • the analysis result-processing characteristic correspondence information 32B stored in the memory 32 includes information representing a result (that is, corresponding to each noise state) that can be obtained as an analysis result of the control unit 36, and the above-described information according to each analysis result.
  • the information is associated with the processing characteristic information indicating the processing characteristics to be set for each unit whose processing characteristics are variable.
  • the control unit 36 reads the corresponding processing characteristic information from the analysis result-processing characteristic correspondence information 32B, and the processing characteristic of each part whose processing characteristic is made variable according to the read processing characteristic information. To change.
  • FIG. 14 is a flowchart showing a specific processing procedure to be executed by the control unit 36.
  • step S101 the external microphone output is monitored for a predetermined time. That is, depending on this monitoring process, the non-speech portion (speech period) of the sound pickup signal from the external microphone 1C is detected.
  • the non-speech part is detected by using, for example, the fact that general environmental noise is (quasi-) stationary compared to speech speech, and monitoring the microphone output within a certain period of time, and the level is low This is done by detecting the period as a non-speech part.
  • step S102 noise analysis is performed in the non-speech detection part. Specifically, frequency analysis is performed on the collected sound signal portion detected as a non-speech portion in the process of step S101.
  • the frequency analysis in step S102 can be realized using BPF (band pass filter), FFT (fast Fourier transform), or the like.
  • step S102 parameter control of each part is performed based on the noise analysis result in step S103. That is, based on the noise analysis result in step S102 and the information content of the analysis result-processing characteristic correspondence information 32B in the memory 32, the processing characteristic of each part whose processing characteristic is made variable as described above is switched. .
  • the sound collection system as the fifth embodiment as described above it is possible to appropriately collect a speech sound with high S / N and high sound quality even if the type of noise around the user changes.
  • FIG. 15 is a diagram illustrating a configuration of a sound collection system according to the sixth embodiment.
  • the S / N / sound quality improving method using the external microphone and the HPF described in the third embodiment and the S / N / sound quality improving method by beam forming described in the second embodiment are used. It is a combination.
  • the mounting portion 1 side corresponds to the Lch and the mounting portion 3 side corresponds to the Rch.
  • the sound collection system according to the sixth embodiment has an external microphone 1 ⁇ / b> C added to the mounting portion 1 and an external microphone 3 ⁇ / b> C added to the mounting portion 3 in comparison with the sound collection system according to the second embodiment.
  • the difference is that a signal processing unit 40 is provided instead of the signal processing unit 20.
  • the external microphone 3C is installed so that sound generated outside the housing can be directly collected in the same manner as the mounting part 1 side.
  • a MEMS microphone is also used for the external microphone 3C.
  • the signal processing unit 40 is the same as the signal processing unit 25 of the third embodiment with respect to the configuration on the Lch side. That is, the microphone amplifier 10, the equalizer 11, the LPF 14, and the delay circuit 28 are provided for the sound collection signal from the internal microphone 1B, and the microphone amplifier 26 and the HPF 27 are provided for the sound collection signal from the external microphone 1C. After that, the addition unit 29 performs addition for the collected sound signals via the respective units.
  • the Rch side has the same configuration as that of the Lch side described above. That is, as shown in the figure, a microphone amplifier 21, an equalizer 43, an LPF 22, and a delay circuit 44 are provided for a sound collection signal from the internal microphone 3B, and a microphone amplifier 41 and a HPF 42 are provided for the sound collection signal from the external microphone 3C. Are added, and the addition unit 45 performs addition for the collected sound signals via the respective units. As a result, the S / N / sound quality improvement effect similar to that described in the second embodiment can be obtained for the Rch-side uttered sound collection signal.
  • the filter characteristics of the equalizer 43 provided on the Rch side, the cutoff frequency of the HPF 42, and the delay time of the delay circuit 44 are Basically, they may be the same as those of the equalizer 11, the HPF 27, and the delay circuit 28, respectively.
  • the signal processing unit 40 is provided with an amplifier 15. Also in this case, the monaural reception signal amplified by the amplifier 15 is supplied to the speaker 1S and the speaker 3S, respectively, as in the case of the second embodiment.
  • the signal processing unit 40 is provided with a beam forming processing unit 23, a noise gate processing unit 12, and a compressor 13 as in the case of the second embodiment.
  • the beamforming processing unit 23 performs a beamforming process based on the Lch-side collected sound signal obtained by the adding unit 29 and the Rch-side collected sound signal obtained by the adding unit 45.
  • the same noise suppression effect as in the beam forming process in the second embodiment can be obtained, and as a result, further S / N improvement can be achieved for the uttered sound collection signal. .
  • the sound collection system according to the present technology is used for calling is exemplified, but the present technology can also be suitably applied to a system for recording the collected speech signal.
  • the sound collection is performed in monaural.
  • the beamforming processing unit 23 in FIG. 15 is omitted, and the output of the addition unit 29 and the output of the addition unit 45 are output independently.
  • the noise gate processing unit 12 and the compressor 13 are provided independently for the output of the adding unit 29 and the output of the adding unit 45, respectively, so that the sound quality of each of the Lch transmission signal and the Rch transmission signal is further improved. It can also be configured.
  • the speakers 1S and 3S are based on the BA type. However, the speakers 1S and 3S can be replaced with those based on the dynamic type or the capacitor type.
  • the system is not particularly limited with respect to the internal microphones 1B and 3B and the external microphones 1C and 3C.
  • the present technology may be configured as follows. (1) At least a part thereof is configured to be insertable into the ear canal part, and a mounting part configured to form a substantially sealed internal space connected to the external auditory canal inside the ear hole part in a mounted state; An internal microphone that is disposed in the internal space of the mounting portion and picks up the uttered sound that is emitted by the wearer and propagates through the ear canal in a state of being mounted on the ear canal portion; A low-frequency extraction filter unit that performs a filtering process for extracting a low-frequency component from the sound collected by the internal microphone, or An ear hole mounting type sound collecting device including any of equalizing processing units for performing high-frequency emphasis type equalizing processing on a sound collected signal from the internal microphone.
  • An external microphone arranged to pick up the external sound of the mounting part; A mid / high range extraction filter unit that performs filtering processing to extract mid / high range components from the sound collected by the external microphone; An adder that adds the collected sound signal filtered by the middle / high-frequency extraction filter unit and the collected sound signal filtered by the low-frequency extraction filter unit (1)
  • the ear hole mounting type sound pickup device according to claim 1.
  • a first internal microphone is disposed as the internal microphone
  • a second internal microphone is disposed as the internal microphone
  • the low-frequency extraction filter unit is Applying the filtering process to the collected sound signal from the first internal microphone and the collected sound signal from the second internal microphone, respectively, Based on the collected sound signal from the first internal microphone subjected to the filtering process by the low-frequency extraction filter unit and the collected sound signal from the second internal microphone subjected to the filtering process by the low-frequency extraction filter unit.
  • the ear hole mounting type sound collecting device further including a beam forming processing unit that performs a beam forming process.
  • (1) to (1) including at least one of a noise gate processing unit that performs noise gate processing on a sound collected signal by the internal microphone and a compressor processing unit that performs compressor processing on the sound collected signal by the internal microphone. 4.
  • the ear canal mounting type sound collecting device according to 4).
  • (6) The ear hole-mounted sound collecting device according to any one of (1) to (5), wherein the filter processing characteristics of the low-frequency extraction filter section are variable.
  • the ear hole-mounted sound collecting device according to any one of (5) to (7), wherein a processing characteristic of at least one of the equalizing processing unit, the noise gate processing unit, or the compressor processing unit is variable.
  • the ear hole-mounted sound collecting device further including a control unit that performs switching control of filter processing characteristics of the low-frequency extraction filter unit according to an operation input.
  • the ear-hole-mounted sound collecting device further including a control unit that performs switching control of filter processing characteristics of the low-frequency extraction filter unit according to a result of noise analysis based on a sound collection signal of external noise. .
  • the control unit The ear hole wearing type according to (10), wherein a non-speech period in which the level of the sound collection signal with respect to the external noise is a predetermined level or less is detected, and the noise analysis is performed based on the sound collection signal in the non-speech period. Sound collection device.
  • At least a part of the mounting part is configured to be insertable into the ear hole part, and the mounting part is configured so that a substantially sealed internal space connected to the ear canal is formed in the ear hole part in a mounted state.
  • the low-frequency extraction filter unit that performs the filtering process of A signal processing apparatus comprising: an equalizing processing unit that performs high-frequency emphasis type equalizing processing on a sound-collected signal from the internal microphone.

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PCT/JP2012/081054 2011-12-08 2012-11-30 耳孔装着型収音装置、信号処理装置、収音方法 Ceased WO2013084810A1 (ja)

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EP12854783.3A EP2790416B1 (en) 2011-12-08 2012-11-30 Earhole attachment-type sound pickup device, signal processing device, and sound pickup method
US14/360,948 US9237392B2 (en) 2011-12-08 2012-11-30 Earhole-wearable sound collection device, signal processing device, and sound collection method
CN201280058984.8A CN103959813B (zh) 2011-12-08 2012-11-30 耳孔可佩戴式声音收集设备，信号处理设备和声音收集方法
US14/992,906 US9918162B2 (en) 2011-12-08 2016-01-11 Processing device and method for improving S/N ratio
US15/883,667 US11070910B2 (en) 2011-12-08 2018-01-30 Processing device and a processing method for voice communication
US17/303,673 US11765497B2 (en) 2011-12-08 2021-06-04 Earhole-wearable sound collection device, signal processing device, and sound collection method
US18/446,275 US12501208B2 (en) 2011-12-08 2023-08-08 Earhole-wearable sound collection device, signal processing device, and sound collection method

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