JP2012008391A - Device and method for changing voice, and confidential communication system for voice information - Google Patents

Abstract

PROBLEM TO BE SOLVED: To provide a device, a method and a system for concealing a content of a voice while suppressing increase in a noise level.SOLUTION: A confidential communication system for voice information has a microphone Mic to receive a voice while a customer 6 in a booth 2 is uttering and to generate a voice signal representing the voice, a SD controller part SD to change the voice signal generated by the microphone Mic, and a speaker SP to convert the voice signal changed by the SD controller part SD into the voice and to output the voice to a neighboring booth 2' where the voice being uttered are being heard. The SD controller part SD includes a partial extraction part to extract a signal of a change subject from the voice signal generated by the microphone Mic based on a waveform of the voice signal, a partial change part to change the signal of the change subject part, and an output part to output the voice signal changed in such a way to the speaker SP.

Description

  The present invention relates to a voice changing device that changes voice, a voice changing method, and a voice information secret talk system including the voice changing device.

  With the enforcement of the Personal Information Protection Law, there is an increasing need to protect conversation information in banks and offices. Conventionally, sound insulation / soundproofing that physically separates the space, BGM / masking system that conceals conversational speech with other noise / music, etc. in an open plan office have been proposed.

For the purpose of concealing voice information,
(1) Masking system that masks the target speech with other stationary noise
(2) Shading system concealed by indoor background noise and air conditioning noise
(3) Sound insulation / sound insulation (the target room is spatially separated and acoustically separated)
Etc. The example (1) attempts to (forcefully) erase the presence of speech, and is positioned as energy masking. This is used, for example, in an open plan office booth or conference room.

  An example of the system (1) is reported in Non-Patent Document 1. There, a dedicated generator or speaker is installed inside the ceiling, etc., and masking sound is generated to conceal the sound. The principle is that it generates music and noise that is not in the way of conversation (contrast with conversation), conceals the contents of speech by reducing so-called S / N, and reduces clarity and intelligibility. Or to conceal the content of the conversation to the extent that it cannot be understood. The system includes a control device (signal processing device), a power amplifier, and the like that control the masking sound to the optimum level according to the conversation level, background noise, and the like.

  In addition, as an example using this technology, noise for masking was radiated from the partition into the booth, and the target space area was limited to the booth to suppress the increase in the noise level in the entire room. There is.

  An example of the system (2) is reported in Non-Patent Document 2. There are reports of “Sound Shading System” that uses indoor background noise itself and air-conditioning noise that is familiar everyday as radiating masking noise. In this system, a speaker is installed at the top of the partition for the purpose of protecting the privacy of conversation, in contrast to a visually interrupting partition for the purpose of ensuring privacy at a bank counter. A masking sound is reproduced from this speaker, thereby preventing the leakage / transmission of conversation contents to a person on the other side of the partition. The sound to be reproduced is the sound generated based on the crowds of the city or the air conditioning noise of the room.

  As an example of the system of (3), there is sound insulation partitioned as a separate room or soundproof partitioned by a partition.

JP 2008-233671 A

KOKUYO Press Release, Sound Masking, October 18, 2006 Akiko Sugimoto, Takahiro Nakamura, Shiro Ise, “Evaluation of Sound Shading System that generates sound field in consideration of ease of conversation and privacy”, Acoustical Society of Japan Spring Meeting 2005, Proceedings, p. 817 The Institute of Electronics, Information and Communication Engineers, Auditory and Speech, 1973, p. 370-371 Iwata, Kobayashi, Takeda, Itakura, “Analysis of phonetic features contained in human speech-like noise”, Journal of the Acoustical Society of Japan, May 1, 1997, 53 (5), p. 337-345

The inventor has recognized the following problems regarding the above-described masking / shading technique.
(I) Since the original sound emits a new sound that is unrelated to the original sound, it is accompanied by a sense of incongruity, and the masker is not always optimal or has the maximum effect corresponding to the original sound.
(II) Noise, that is, a masking sound can be heard even when the sound is not generated. Therefore, the noise level in the indoor space can be reliably increased.
(III) By emitting another sound (noise / music) unrelated to conversation, it is possible to give a sense of incongruity to a speaker, a talker, and other people in the room.
(IV) Concealment of speech information includes the basic problem that it is difficult to succeed with noise and BGM due to the auditory nature of distinguishing and recognizing different things (envelope and spectrum are similar) Audio waveforms are more difficult to distinguish for auditory recognition).

  As for (I), the relative level of noise necessary for completely masking the original voice is empirically about 15 dB (see Non-Patent Document 3). From this point of view, the method of concealing sound by flowing noise and music requires much louder noise and music than the original sound, and whether it is masking or shading, the room noise level Can be greatly increased.

Regarding (II), there is a sense of incongruity that a sound is produced even when there is no utterance. In the first place, playing noise and music when there is no utterance is useless from the viewpoint of concealing conversation content. Not only is it wasteful, but it also _{increases the} room equivalent noise level (L _Aeq : A-weighted equivalent sound level = the root mean square sound pressure level of the audio signal corrected with the A characteristic, that is, the average noise level). Can result. Even when “HSL noise (Human Speech-like noise)” (see Non-Patent Document 4) created from music or voice is used instead of noise, it is difficult to distinguish from general BGM.

  In addition, the approach (3) is considerably large in cost and hinders a feeling of opening, and is not suitable for use in an open plan office or the like.

  Further, in the sound masking system described in Patent Document 1, a method is described in which the speech speed of an input sound (voice) is analyzed, divided and processed by a frame length corresponding to this, and the processed speech is synthesized. However, since this system “stores the input sound (voice) temporarily in about 2 seconds and performs a series of processing”, the processed sound is generated from the past sound that is different from the sound that is the masking target. . Therefore, the relevance between the processed voice and the voice targeted for masking is low, and the masking effect is not sufficient.

  The present invention has been made in view of these problems, and an object of the present invention is to provide a technique for concealing audio content while suppressing an increase in noise level and listener discomfort.

  One embodiment of the present invention relates to a sound changing device. The voice changing device is configured to extract a signal of a change target part from a voice signal representing a voice being uttered based on a waveform of the voice signal, and change a signal of the change target part extracted by the partial extraction part. And an output unit that outputs the signal of the change target portion changed by the partial change unit to a voice output unit capable of outputting voice to an area where the voice being spoken is received.

  According to this aspect, it is possible to determine the part to be changed in the audio signal based on the waveform of the audio signal.

  Another aspect of the present invention is a speech information secret talk system. The voice information secret speech system is modified by a sound collecting unit that receives a voice being uttered and generates a voice signal representing the voice, a voice changing device that changes a voice signal generated by the sound collecting unit, and a voice changing device. Voice output means for converting the received voice signal into a voice and outputting the voice signal to a region where the voice being spoken is received. The sound changing device includes a partial extraction unit that extracts a signal of a change target portion based on a waveform of the sound signal from the sound signal generated by the sound collecting unit, and changes the signal of the change target portion extracted by the partial extraction unit. And an output unit that outputs the signal of the change target portion changed by the partial change unit to the audio output unit.

  It should be noted that any combination of the above-described constituent elements, or those obtained by replacing the constituent elements and expressions of the present invention with each other between apparatuses, methods, systems, computer programs, recording media storing computer programs, and the like are also included in the present invention. It is effective as an embodiment of

  ADVANTAGE OF THE INVENTION According to this invention, the content of an audio | voice can be concealed, suppressing the increase in a noise level and a listener's discomfort.

It is explanatory drawing which divides into a category the conventional approach regarding masking, and the approach which concerns on embodiment. It is a perspective view which shows typically the booth provided with the audio | voice information secret talk system which concerns on embodiment. It is a block diagram which shows typically the function and structure of the audio | voice information confidential system of FIG. It is a side view which shows the structure of the IT partition of FIG. It is a block diagram which shows the function and structure of the SD controller part of FIG. It is a data structure figure which shows the consonant library of FIG. It is a wave form diagram which shows the waveform of the audio | voice signal showing an example of a maskee. 6 is a waveform diagram showing a waveform of an audio signal generated by processing the audio signal of FIG. 7 in the SD controller unit of FIG. 5 in a consonant only replacement mode. It is explanatory drawing for demonstrating the determination criteria of the signal of the change object part in a 1st determination part. It is a wave form diagram which shows the waveform of the audio | voice signal showing the masker and the time rotation process masker in a listener position. It is a flowchart which shows a series of processes in the audio | voice information confidential system of FIG. It is a block diagram which shows typically the function and structure of the audio | voice information confidential system which concerns on a 1st modification. It is a block diagram which shows typically the function and structure of the audio | voice information confidential system which concerns on a 2nd modification.

  The present invention will be described below based on preferred embodiments with reference to the drawings. The same or equivalent components, members, and processes shown in the drawings are denoted by the same reference numerals, and repeated descriptions are omitted as appropriate.

  Particularly in offices and the like, it is desirable to conceal only the voice information, that is, the voice content without impairing the openness and smoothness of communication of the open plan space. However, the conventional technology using BGM and masking basically adds a sound that is different in nature from the original sound and created in a separate process without any relation to the original sound. There was a dislike that would raise. The embodiment of the present invention changes the structure of the audio signal itself collected by a microphone or the like substantially in real time, thereby improving the conversation content without increasing the background noise in the room, ideally the conversation content. Hides only, and realizes a smooth and comfortable secret story environment.

  FIG. 1 is an explanatory diagram showing a conventional approach related to masking and an approach according to an embodiment divided into categories. (A) is SR (Sound Reinforcement) / PA (Public Address) using electroacoustics. These are conventional technologies that increase the volume and clarity and “make them sound better”. (F) is Sound Insulation, which is a conventional technique for acoustically separating a space and making it “not audible” as much as possible. On the other hand, the approach according to the embodiment is SD (Speech Deformation) of (e). By processing the original voice of the conversation person and outputting it in near real time, the conversation contents are not heard but not heard. It is a kind of voice information disruption (hearing) technique that makes it “unknown”. Further, (b) EM, (c) SS, and (d) IM according to the prior art can increase the noise level in the room or the target space region to increase the unpleasantness and discomfort, SD hardly causes an increase in noise level.

The main standpoint of the embodiment of the present invention is the recognition of the present inventor that language recognition and understanding depend largely on the consonant part of speech, particularly in the case of Japanese. If this consonant part changes, for example, “KUMO” becomes “RUTO” and cannot be understood as words.
Also, based on the fact that auditory speech recognition (HSR: Human Speech Recognition) is more dependent on articulation such as envelope transitions than the carrier of the audio signal, it processes the “sounds of the envelope” of the original speech. When time reversal or time rotation is performed as a target unit, since the spectrum and the envelope shape are similar to the original speech, the speech information disturbance functions effectively.

In the embodiment of the present invention, paying attention to such aspects of speech recognition / understanding, the consonant part of the original speech is changed / deleted / replaced in a certain mode. Since the processing of the consonant part is the main, the increase in the sound pressure level (volume) is small compared to the original voice. Further, in order to further reduce the overall volume of the processed voice (hereinafter referred to as a masker) added to the original voice (hereinafter referred to as a maskee), the following combination / ingenuity is possible.
(I) In generating a masker, the vowel part is replaced with silence, and only the processed consonant part is output at the original timing.
(Ii) ANC (Active Noise Control) or parameter-fixed PNC (Passive Noise Control) technology is used in combination to enhance the masker's information hiding effect.

FIG. 2 is a perspective view schematically showing the booth 2 in which the audio information secret system 100 according to the embodiment is provided. FIG. 3 is a block diagram schematically showing the function and configuration of the speech information secret system 100 of FIG.
The voice information secret story system 100 is provided in a booth 2 partitioned by a simple partition, such as a bank consultation counter. The audio information secret system 100 includes a microphone Mic, an SD controller unit SD, two power amplifiers PA, and two speakers SP. The speaker SP and the SD controller unit SD may be incorporated in the IT partition 4 that visually separates the booths.

  A customer 6 who has a conversation with a counselor is a speaker. The speaker's maskee H ′ (t) is collected by a microphone Mic provided at or near the counter portion. The maskee H ′ (t) collected by the microphone Mic is converted into an audio signal and sent to the SD controller unit SD. This audio signal is changed, deleted, replaced, or reversed / rotated in time by the SD controller unit SD. The audio signal that has undergone processing in the SD controller section SD is output as a masker H (t) from the speaker SP to the left and right adjacent booths 2 'via the power amplifier PA.

  Since Muskie H '(t) goes around the air in the adjacent booth 2', the voice being spoken by the customer 6 is received by a listener 8 (a person different from the customer 6) in the adjacent booth 2 '. sell. However, in the present embodiment, the masky H ′ (t) that leaks through the air is combined with the masker H (t) and reaches the listener 8 in the adjacent booth 2 ′. Therefore, the listener 8 cannot understand the content of the conversation included in the maskee H ′ (t) due to the disturbance caused by the masker H (t).

  The speaker SP outputs a masker H (t) toward the adjacent booth 2 'adjacent to the booth 2 where the SD controller unit SD and the microphone Mic are installed. Here, the adjacent booth 2 ′ is an area where the muskey H ′ (t) leaking around the air is received. That is, the masker H (t) and the masker H (t) are output from the speaker SP so that the masker H (t) and the masker H (t) reach the listener 8 substantially in real time. The main body that guarantees the real-time property may be the SD controller unit SD or the speaker SP, but in the following, the SD controller unit SD performs the real-time property of the maskee H ′ (t) and the masker H (t). A case where an audio signal is processed in consideration of the above will be described.

  FIG. 4 is a side view showing the configuration of the IT partition 4 of FIG. The IT partition 4 has a laminated structure in which a first sound absorbing layer 42, a sound insulating layer 44, and a second sound absorbing layer 46 are laminated in this order. Each of the first sound absorbing layer 42 and the second sound absorbing layer 46 is a glass wool layer having a thickness of 20 mm. The sound insulation layer 44 is a gypsum board having a thickness of 12 mm.

  FIG. 5 is a block diagram showing the function and configuration of the SD controller unit SD of FIG. Each block shown here can be realized in hardware by an element such as a CPU (central processing unit) or a mechanical device, and in software by a computer program or the like. Describes functional blocks realized by collaboration. Accordingly, it is understood by those skilled in the art who have touched this specification that these functional blocks can be realized in various forms by a combination of hardware and software.

  The SD controller unit SD includes a storage device 10, an A / D unit 20, a partial extraction unit 30, a partial change unit 90, an output unit 72, a noise generation unit 80, a consonant library update unit 82, and a vowel library. An update unit 84. The storage device 10 includes a consonant library 12, a vowel library 14, and a common library 16. The partial extraction unit 30 includes a phoneme extraction unit 38, a substantially single mountain extraction unit 52, and a random extraction unit 60. The phoneme extraction unit 38 includes a speech discrimination unit 36, a consonant extraction unit 32, and a vowel extraction unit 34. The approximately one mountain extraction unit 52 includes a square sound pressure acquisition unit 54, a low-pass filter 56, and a first determination unit 58. The random extraction unit 60 includes a signal division unit 62 and a second determination unit 64. The partial changing unit 90 includes a consonant processing unit 40, a vowel processing unit 50, and a time processing unit 66. The output unit 72 includes a delay adjustment unit 68 and a D / A unit 70.

  The consonant library 12 stores waveform data for each type of consonant part. The vowel library 14 stores waveform data for each type of vowel part. The common library 16 stores predetermined sample waveform data for each type of consonant part. The sample waveform data of the consonant part stored in the common library 16 is classified into male, female, child, adult and the like.

  The partial extraction unit 30 extracts the signal of the change target portion from the audio signal A / D converted by the A / D unit 20 based on the waveform of the audio signal. The partial change unit 90 changes the signal of the change target portion extracted by the partial extraction unit 30. The output unit 72 D / A converts the signal of the change target portion changed by the partial change unit 90 and outputs the signal to the speaker SP.

  The SD controller unit SD has at least three operation modes: a consonant only replacement mode, a consonant vowel replacement mode, and a real time mode. The function of the block related to each operation mode will be described below.

(1) Consonant-only replacement mode The masky H ′ (t) collected by the microphone Mic is converted into an audio signal, and the audio signal is input to the A / D unit 20 via a microphone amplifier (not shown). The A / D unit 20 converts an audio signal that is an analog signal into a digital signal. The voice discriminating unit 36 discriminates a consonant part and a vowel part of the voice signal by comparing the waveform of the voice signal digitized by the A / D unit 20 with a past speech voice waveform. The consonant extraction unit 32 extracts the signal of the consonant part using the determination result.

  The consonant library update unit 82 stores the waveform data of the consonant part signal extracted by the consonant extraction unit 32 in the consonant library 12 for each type. Here, the classification of the consonant part is performed based on its duration, spectrum, statistical processing, and the like. In this way, the waveform data of the consonant signal stored in the consonant library 12 is gradually replaced with higher accuracy from the start of the conversation by sequential processing.

  Based on the signal of the consonant part extracted by the consonant extraction unit 32, the noise generation unit 80 generates a sound whose spectrum overlaps or is different.

  The consonant processing unit 40 processes the signal of the consonant part extracted by the consonant extraction unit 32 in the audio signal. The consonant processing unit 40 replaces the signal of the consonant part extracted by the consonant extraction unit 32 with the signal of another consonant part selected from the consonant library 12 and having substantially the same length. When there are a plurality of replacement candidates, the consonant processing unit 40 performs replacement randomly and so that each combination has a substantially equal probability. Here, as an example of the length of the consonant part, the duration of the consonant part corresponding to “s” is relatively long, and the duration of the consonant part corresponding to “t” or “p” is short. Sometimes.

  The consonant processing unit 40 replaces the consonant part signal extracted by the consonant extraction unit 32 with the consonant noise generated by the noise generation unit 80 instead of replacing the consonant part signal using the consonant library 12. May be. In this case, the randomness of the synthesized speech between the maskee H ′ (t) and the masker H (t) is further increased. The consonant processing unit 40 may delete the consonant part signal extracted by the consonant extracting unit 32 instead of replacing the consonant part signal using the consonant library 12.

  There is a possibility that the consonant portion collected from the voice of the utterer is not sufficiently accumulated in the consonant library 12 for several seconds to several tens of seconds (hereinafter referred to as an utterance start period) from the start of the utterance. Therefore, during this utterance start period, the consonant processing unit 40 selects a corresponding consonant part signal from the common library 16 and replaces it with the consonant part signal extracted by the consonant extraction unit 32. Alternatively, the consonant processing unit 40 replaces the consonant part signal extracted by the consonant extraction unit 32 with the consonant noise generated by the noise generation unit 80 during the utterance start period. Alternatively, during the utterance start period, the consonant processing unit 40 inverts the signal of the consonant part extracted by the consonant extraction unit 32 in the time direction.

  These consonant partial modification algorithms used during the utterance start period are less natural than using the consonant library 12 of the speaker himself. However, since it is only a short time after the start of utterance, it does not matter so much.

  The D / A unit 70 converts the audio signal processed by the consonant processing unit 40 into an analog audio signal for driving the speaker SP and outputs the analog audio signal to the power amplifier PA. In particular, the D / A unit 70 converts an audio signal including a signal of the consonant part replaced by the consonant processing unit 40 and an unmodified vowel part signal corresponding to the consonant part into an analog signal and outputs the analog signal. .

Note that the time from when the musky H ′ (t) is collected by the microphone Mic until it is processed by the SD controller unit SD and the corresponding masker H (t) is output from the speaker SP, that is, the SD processing time T _SD is T + t. It is supposed to be within. Here, T is the time from when Muskie H '(t) is issued until it reaches the listener 8, and t is noticeable when the Muskie H' (t) and Masker H (t) are at the listener 8 position. This is the delay time that does not generate a simple echo, or the maximum delay time that the synthesized speech that reaches the listener 8 becomes unintelligible for the listener 8. Although the specific value of t is determined by experiment, it is typically about several hundred ms.

In order to conceal information by synthesizing Masky H '(t) and Masker H (t) at the listener's 8 position, the SD processing in the SD controller unit SD must be performed in real time or near real time as described above. I must. Due to the presence of this time constraint, that is, the SD processing time T _SD must be shorter than T + t, which is a short time, the accuracy of processing such as extraction, replacement and inversion of consonant signal must be sacrificed. In some cases. However, the purpose of this embodiment is to reduce the intelligibility and intelligibility of speech, and the accuracy of the assumed / scheduled processing itself is not the purpose. Therefore, in this embodiment, if the condition that it becomes difficult to understand the meaning content of the maskee H ′ (t) due to the superposition of the maskers H (t), the processing accuracy does not become a big problem. This is because there are an infinite number of “conditions that make it difficult to understand the semantic content”.

(2) Vowel replacement mode In this mode, the vowel part is also changed in addition to the above-described change of the consonant part. The vowel extraction unit 34 extracts the vowel part signal from the voice signal from which the consonant part signal is extracted by the consonant extraction unit 32.

  The vowel library updating unit 84 stores the waveform data of the vowel part signal extracted by the vowel extraction unit 34 in the vowel library 14 for each type. Here, the vowel part is classified based on its duration, spectrum, statistical processing, and the like. In this way, the waveform data of the signal of the vowel part stored in the vowel library 14 is gradually replaced with one having higher accuracy from the start of the conversation by sequential processing.

  The noise generation unit 80 generates vowel noise having a spectrum similar to that of the vowel part signal extracted by the vowel extraction unit 34.

  The vowel processing unit 50 processes the signal of the vowel part extracted by the vowel extraction unit 34 out of the voice signal after the signal of the consonant part is processed by the consonant processing unit 40. In particular, when it is necessary to suppress an increase in noise level as much as possible, the vowel processing unit 50 replaces the vowel part extracted by the vowel extraction part 34 with a silent part. In this case, the masker H (t) output via the D / A unit 70 and the speaker SP has a structure having a silent part sandwiched between a consonant part and a consonant part. That is, the consonant part of the masker H (t) is connected to the vowel part of the synchronized masky H ′ (t) to form one phoneme. As a result, the overall sound volume is reduced by the vowel part silenced by the masker H (t), and the indoor noise level is also reduced.

  The vowel processing unit 50 may perform library-based replacement instead of replacing the vowel part with a silent part. In other words, the vowel processing unit 50 may replace the signal of the vowel part extracted by the vowel extraction unit 34 with the signal of another vowel part selected from the vowel library 14. When there are a plurality of replacement candidates, the vowel processing unit 50 performs replacement so that each combination has a substantially equal probability. The vowel part changing algorithm in the utterance start period is the same as that of the consonant part.

  Alternatively, the vowel processing unit 50 may replace the vowel part signal extracted by the vowel processing unit 50 with the vowel noise generated by the noise generation unit 80 instead of replacing the vowel part with a silent part. In this case, the randomness of the synthesized speech of the maskee H ′ (t) and the masker H (t) is further increased.

  Further, the order of processing of consonant vowels, that is, the order of processing in the consonant processing unit 40 and processing in the vowel processing unit 50 may be switched.

  FIG. 6 is a data structure diagram showing the consonant library 12. The consonant library 12 stores a consonant 112 as a phoneme and waveform data 114 of the consonant in association with each other. The vowel library 14 and the common library 16 also have the same data structure as the consonant library 12.

  FIG. 7 is a waveform diagram showing a waveform of an audio signal representing an example of the maskee H ′ (t). The waveform in FIG. 7 is obtained by converting the original voice “ANO KARETOWA SO-TONAGAINDAYONE ZITSUWA” into a voice signal with the microphone Mic. The vertical axis in FIG. 7 represents signal intensity in arbitrary units, and the horizontal axis represents time. In FIG. 7, each region divided by vertical broken lines corresponds to a phoneme, and the corresponding phoneme is clearly shown in Roman letters. “-” Represents a voice pause unit. Envelope 102 is shown as a solid line. Here, the envelope is obtained by multiplying a voice sample by a time constant of several tens of msec in the square sound pressure region and taking a square root.

  Table 1 shows vowels, consonants, and silences in FIG. A certain time before the start of voice is defined as the time origin (t = 0).

なお、子音、母音、無音の別は、エネルギやゼロ交差数、ＰＡＲＣＯＲ（PARtial auto-CORrelation）の第１係数（スペクトル傾斜）などにより判別することが可能である。

The distinction between consonants, vowels, and silence can be determined by energy, the number of zero crossings, the first coefficient (spectral slope) of PARCOR (PARtial auto-CORrelation), and the like.

FIG. 8 is a waveform diagram showing a waveform of an audio signal generated by processing the audio signal of FIG. 7 in the consonant only replacement mode in the SD controller unit SD. This is a consonant part in which the consonant part indicated by the section 104 is replaced. For these replacements, the cut-out time length and re-insertion level (dB) are adjusted.
The replacement envelope 106 is indicated by a solid line. When the envelope curve 102 in FIG. 7 is compared with the envelope curve 106 in FIG. 8, it can be seen that there is not much change. In other words, there is not much change in voice intonation and intonation. However, when the audio signal of FIG. 8 is converted into sound by the speaker SP and output as a masker H (t), the listener 8 site synthesizes and hears the maskey H ′ (t) and the masker H (t), Its meaning is difficult to understand. In other words, it is often “I don't know” (may be heard by other sounds).

Returning to FIG.
(3) Real-time mode Musky H ′ (t) collected by the microphone Mic is converted into an audio signal, and the audio signal is input to the A / D unit 20 via a microphone amplifier (not shown). The A / D unit 20 converts an audio signal that is an analog signal into digital data. The audio signal digitized by the A / D unit 20 is digital data in which a voltage value corresponding to the magnitude of sound pressure is associated with time, for example.

  The partial extraction unit 30 extracts the signal of the change target portion from the audio signal digitized by the A / D unit 20. The partial extraction unit 30 may extract a consonant part signal as a change target part signal. Or the partial extraction part 30 may extract the signal of a vowel part as a signal of a change object part. The extraction of the consonant part and the vowel part is as described above.

  Alternatively, the partial extraction unit 30 may extract a group of signals determined based on the shape of the envelope of the audio signal as the signal of the change target portion. Alternatively, the partial extraction unit 30 may divide the audio signal by a period having a random length and extract a signal corresponding to one section after the division as a signal of the change target part.

  A case will be described in which the partial extraction unit 30 extracts a group of signals determined based on the shape of the envelope of the audio signal as the signal of the change target portion. The approximately one mountain extracting unit 52 acquires data indicating the envelope of the audio signal. This data is digital data in which a voltage value corresponding to the size of the envelope is associated with time, for example. Hereinafter, data indicating an envelope is simply referred to as an envelope.

  The squared sound pressure acquisition unit 54 acquires the squared sound pressure waveform of the audio signal digitized by the A / D unit 20. The squared sound pressure acquisition unit 54 squares the audio signal and obtains a squared sound pressure waveform by multiplying by a predetermined coefficient as necessary.

The low pass filter 56 averages the squared sound pressure waveform acquired by the squared sound pressure acquisition unit 54 with a time constant of several msec to several hundred msec. That is, the low pass filter 56 performs low pass filter processing on the squared sound pressure waveform. Thereby, a change faster than the time constant is removed from the squared sound pressure waveform, and a smooth waveform is obtained. In this embodiment, this smooth waveform is the envelope of the audio signal. It should be understood by those skilled in the art who have touched this specification that the envelope of the audio signal may be obtained by other methods. In the present embodiment, the envelope is data indicating a change in average energy (amplitude) of the audio signal in a broad sense.
The low pass filter 56 takes the square root of the low pass filtered data if necessary.

  The first determination unit 58 detects an ascending portion continuously rising from several dB to several tens dB, for example, 5 dB or more, from the envelope of the audio signal obtained by the low-pass filter 56. Next, the first determination unit 58 detects a descending portion that descends continuously several dB to several tens dB, for example, 5 dB or more after the ascending portion. The 1st determination part 58 determines the audio | voice signal between a raise part and the fall part corresponding to it as a signal of a change object part. In many cases, the envelope of the signal of the change target portion determined in this way is approximately one mountain.

  FIG. 9 is an explanatory diagram for explaining a determination criterion for the signal of the change target portion in the first determination unit 58. FIG. 9A is an explanatory diagram for explaining a case where the signal of the change target portion is determined based on the detection of the rising portion and the falling portion in the first determination unit 58. FIG. 9A shows an exemplary audio signal waveform 211 and its envelope 208. The first determination unit 58 detects the rising portion 202 based on the rate of change of the envelope 208. Next, the first determination unit 58 detects the descending portion 204 after the ascending portion 202. The first determination unit 58 uses the audio signal of the section 206 (the section sandwiched between the time t1 before the peak 203 and the time t2 after the peak 203) sandwiched between the rising portion 202 and the descending portion 204 as the change target portion. Determine as a signal.

  Note that the first determination unit 58 may determine the signal of the change target portion by another method. For example, the first determination unit 58 may detect a portion where the envelope is swelled and determine a sound signal corresponding to the portion as a signal of the change target portion. Or the 1st determination part 58 may detect the peak of an envelope, and may determine the audio | voice signal of the area which has a predetermined length before and behind as a signal of a change object part. Or the 1st determination part 58 may determine the audio | voice signal of the continuous area where an envelope exceeds a predetermined level as a signal of a change object part.

  FIG. 9B is an explanatory diagram for explaining a case where the signal of the change target portion is determined based on the detection of the peak in the first determination unit 58. FIG. 9B shows an exemplary sound signal waveform 212 and its envelope 214. The first determination unit 58 detects the peak 216 of the envelope 214. The first determination unit 58 determines the audio signal of the section 218 having a predetermined length before and after the peak 216 as the signal to be changed.

  FIG. 9C is an explanatory diagram for explaining a case where the signal of the change target portion is determined based on the envelope level in the first determination unit 58. FIG. 9C shows an exemplary audio signal waveform 220 and its envelope 222. The first determination unit 58 detects a continuous section 226 in which the envelope 222 exceeds a predetermined level 224, and determines the audio signal in the section 226 as a signal to be changed. In this case, depending on how to obtain a predetermined level, the signal of the change target portion may include two or more peaks.

  As described above, various methods for determining the signal of the change target portion are conceivable. Such a large number of options is preferable in the sense that it provides a great degree of freedom for making the concealment of conversation contents by SD more effective.

  In addition, what can be said through these various determination methods is that a group of signals is determined based on the waveform of an audio signal, particularly based on its statistical properties, and the group of signals thus determined is That is, it is determined as a signal of the part to be changed. That is, the change target portion is adaptively determined according to the incoming audio signal. In this case, according to the experience of the present inventor as a person skilled in the art and preliminary experiments, it has been found that the conversation content disturbance effect is higher than, for example, the case where audio signals are cut out at predetermined intervals. . In particular, according to an experiment conducted by the present inventor, when approximately one peak of an envelope is extracted as a change unit, the disturbance effect is more effective than, for example, cutting out at a constant period or using a consonant or vowel as a change unit. It was found to be expensive.

Returning to FIG.
The first determination unit 58 outputs a portion of the audio signal that has not been determined as the signal to be changed to the delay adjustment unit 68.

A case will be described in which the partial extraction unit 30 divides the audio signal by a period having a random length and extracts a signal corresponding to one section after the division as a signal of the change target part.
The signal dividing unit 62 divides the audio signal digitized by the A / D unit 20 in a period having a random length. The length of the period varies between several tens of milliseconds to several hundreds of milliseconds. Alternatively, the length of the period varies within a range of ± several tens of% to several hundreds of% with respect to a certain period. For example, the length of the period changes as follows: 11 msec, 10 msec, 12 msec,.

The 2nd determination part 64 determines the signal corresponding to one of the periods divided | segmented by the signal division part 62 among audio | voice signals as a signal of a change object part. The second determination unit 64 may select all the divided periods as the change target part, or may select every other period as the change target part, for example. In the latter case, the second determination unit 64 outputs the audio signal of the part corresponding to the period not selected as the change target part to the delay adjustment unit 68.
In this case, since the randomness is added to the length of the period, the naturalness of the masker H (t) is improved.

  The time processing unit 66 processes the signal of the change target portion extracted by the partial extraction unit 30 based on the waveform along the time axis. The time processing unit 66 performs time reversal or time rotation on the signal of the change target portion.

Regarding the time inversion, the time processing unit 66 inverts the extracted signal of the change target portion with respect to time. That is, the time processing unit 66 generates a signal obtained by reversing the time from the signal of the change target portion. More specifically, the time processing unit 66 at the time t _i (0 ≦ i ≦ N, t ₀ <t ₁ <... <T _N , N is a natural number, t ₀ ≡0) of the signal to be changed. A function h (f (t _i )) = f (t _N −t _i ) is applied to the voltage value f (t _i ). As a result, the waveform of the signal of the change target portion that has undergone the time reversal process in the time processing unit 66 has a shape that is folded back with respect to a line that passes through the center of the original waveform and is perpendicular to the time axis.

  Regarding the time rotation, the time processing unit 66 rotates the waveform along the time axis of the extracted signal of the change target portion. More specifically, the time processing unit 66 performs time reversal on the signal of the change target portion as described above. In addition, the time processing unit 66 inverts the sign of the signal of the change target portion subjected to the time inversion. As a result, the waveform of the signal of the change target portion that has undergone the time rotation processing in the time processing unit 66 has a shape obtained by rotating the original waveform by 180 degrees with respect to the center on the time axis.

  The output unit 72 acquires from the time processing unit 66 a signal of the change target portion that has been subjected to time reversal or time rotation processing, and receives from the partial extraction unit 30 a signal that is not the change target portion. The output unit 72 converts them into analog signals and outputs them to the speaker SP via the power amplifier PA.

  The delay adjusting unit 68 generates an output audio signal to be output by connecting the signal of the change target portion that has been subjected to the time inversion or time rotation processing and the signal that is not the change target portion. The delay adjustment unit 68 adjusts the timing at which the output audio signal is output from the output unit 72 according to the time required for propagation of the maskee H ′ (t). In particular, the delay adjustment unit 68 gives a predetermined delay to the output audio signal. This delay is within a range where the masker H (t) delay with respect to the masker H '(t) at the listener 8 position is such that the masker H' (t) and the masker H (t) can be said to be substantially real time. Set to fit.

  The fact that the maskee H '(t) and the masker H (t) are substantially in real time means that, for example, the maskee H' (t) and the masker H (t) are at least partially overlapped in the adjacent booth 2 '. It is to be. Alternatively, the signal of the change target portion output from the output unit 72 is converted into sound by the speaker SP, and the converted sound is received while the maskee H ′ (t) is received in the adjacent booth 2 ′. It is to be output to the adjacent booth 2 ′. Alternatively, the signal of the change target portion output from the output unit 72 is converted into sound by the speaker SP, and the converted sound is adjacent to the portion of the maskee H ′ (t) corresponding to the signal of the change target portion. This is to be output to the adjacent booth 2 ′ while listening in the booth 2 ′. In other words, the portion of the maskee H ′ (t) corresponding to the signal of the change target portion and the portion of the masker H (t) corresponding to the signal of the change target portion are at least partially in the adjacent booth 2 ′. It is to superimpose on.

  When the voice information secret system 100 is introduced, the positions of the microphone Mic and the speaker SP are determined, and the position of the assumed customer 6 and the assumed position of the listener 8 are also determined to some extent. In addition, the processing time in the SD controller unit SD can be estimated to some extent. Therefore, at the time of introducing the voice information secret talk system 100, the propagation time of the maskee H ′ (t) and the propagation time of the masker H (t) from the customer 6 to the listener 8 can be estimated to some extent. The delay in the delay adjusting unit 68 is set by calculating backward from a desired value of the delay of the masker H (t) with respect to the maskee H ′ (t) at the listener 8 position.

  If the masker H (t) has a large delay with respect to the maskee H ′ (t), echoes or reverberations may occur at the listener 8 position. Therefore, the delay adjusting unit 68 sets a delay for the output audio signal such that the delay of the masker H (t) with respect to the maskee H ′ (t) at the listener 8 position is a value that does not cause such a sense of incongruity. give. Although this delay is determined by experiment, it is typically several hundred msec or less.

  Further, depending on the positional relationship between the microphone Mic, the speaker SP, the customer 6, and the listener 8, the masker H (t) is received much later than the maskee H '(t) when the delay adjusting unit 68 does not apply a delay. The position of the listener 8 may be reached. In this case, in order to conceal the information by synthesizing the maskee H '(t) and the masker H (t) in the real time in the listener 8 position, the SD processing time in the SD controller unit SD is shortened. Must. In some cases, the accuracy of time processing must be sacrificed due to the existence of this time constraint, that is, the SD processing time must be shortened. However, the purpose of this embodiment is to reduce the intelligibility and intelligibility of speech, and the accuracy of the assumed / scheduled processing itself is not the purpose. Therefore, in this embodiment, if the condition that it becomes difficult to understand the meaning content of the maskee H ′ (t) due to the superposition of the maskers H (t), the processing accuracy does not become a big problem. This is because there are an infinite number of “conditions that make it difficult to understand the semantic content”.

  The D / A unit 70 converts the output audio signal provided with the delay by the delay adjusting unit 68 into an analog audio signal for driving the speaker SP and outputs the analog audio signal to the power amplifier PA.

  FIG. 10 is a waveform diagram showing the waveform of an audio signal representing the maskee H ′ (t) and the time-rotated masker H (t) at the listener 8 position. FIG. 10A is a waveform diagram showing the waveform of an audio signal representing the maskee H ′ (t). The waveform in FIG. 10A is obtained by converting the original voice into a voice signal using the microphone Mic. In FIG. 10A, the vertical axis represents signal intensity in arbitrary units, and the horizontal axis represents time. FIG. 10B is a waveform diagram showing a waveform of an audio signal generated by subjecting the audio signal of FIG. 10A to time rotation in approximately one mountain unit in the SD controller unit SD. For example, the SD controller unit SD extracts approximately one peak audio signal indicated by a circle 150 in FIG. 10A as a signal to be changed, performs time rotation on the approximately one peak audio signal, and performs FIG. An audio signal indicated by a circle 152 in (b) is generated and output.

  When the envelope of FIG. 10A is compared with the envelope of FIG. 10B, it can be seen that there is not much change. In other words, there is not much change in voice intonation and intonation. However, when the audio signal of FIG. 10B is converted into audio by the speaker SP and output as a masker H (t), the masker H '(t) and the masker H (t) are synthesized at the listener 8 site. The meaning is difficult to understand. In other words, it is often "I don't know".

  FIG. 11 is a flowchart showing a series of processes in the speech information secret system 100. The microphone Mic collects the maskee H ′ (t) and generates an audio signal (step 302). The A / D unit 20 acquires an audio signal representing the maskee H ′ (t) from the microphone Mic (step 304). The partial extraction unit 30 extracts the signal of the change target portion from the audio signal acquired and A / D converted by the A / D unit 20 based on the waveform of the audio signal (step 306). The partial change unit 90 changes the signal of the change target portion extracted by the partial extraction unit 30 (step 308). The output unit 72 outputs the signal of the change target portion changed by the partial change unit 90 to the speaker SP (step 310). The loudspeaker SP converts the received signal into a voice to obtain a masker H (t), and outputs the masker H (t) to the adjacent booth 2 'where the maskee H' (t) is being listened to (step 312).

  The operation of the speech information secret system 100 having the above configuration will be described. Consider a case in which a customer 6 sits in a bank booth 2 and consults with a bank counselor about, for example, a loan. At this time, it is assumed that there is a listener 8 in the adjacent booth 2 ′ next to the booth 2 and an application for opening an account is being made. Customer 6 explains the circumstances of applying for a loan, such as the worsening of the cash flow of his business. Of course, it is better for the listener 8 not to leak such a story. In particular, in the speech information secret speech system 100 according to the present embodiment, a consonant signal converted from the speech of the customer 6 or a time rotation is applied. Since the received information reaches the listener 8, the listener 8 cannot understand the utterance content of the customer 6. In addition, when there is no utterance by the customer 6, there is substantially no output from the speaker SP to the adjacent booth 2 ', so that the noise level in the adjacent booth 2' is not increased unnecessarily.

  In the above-described embodiment, examples of the storage device 10 are a hard disk and a memory. In addition, based on the description of the present specification, each block is stored in a CPU (not shown), an installed application program module, a system program module, a memory that temporarily stores data read from the hard disk, or the like. It will be understood by those skilled in the art who have touched this specification that it can be realized.

  According to the speech information secret system 100 according to the present embodiment, the following operational effects can be obtained.

(1) According to the speech information secret system 100 according to the present embodiment, the content, that is, the information included in the conversation speech is concealed instead of concealing or deleting the presence of the conversation itself. In this regard, the inventor has recognized the following.
In open-plan offices, bank counters, and securities company lobby counters, especially at customer service counters that are partitioned by simple partitions, concealing the content of a conversation is possible if the contents of the conversation cannot be understood by anyone other than the person who is speaking. The point serves its purpose well. In other words, the voice itself may be heard as long as the content of the conversation is not leaked. Rather, when the presence of a speaker can be visually recognized, it is more natural to preserve the speech spectrum and envelope (sound quality, intonation, and intonation). The voice information secret system 100 according to the present embodiment corresponds to the above viewpoints and needs, and conceals conversation contents in a more natural manner.

  (2) When the consonant part is extracted by the partial extraction unit 30, the masker H (t) is created based on the utterance of the speaker's own masque H '(t), focusing on the consonant part, and in parallel with the original speech. Output from the speaker. Therefore, particularly in the consonant-only replacement mode, the spectrum and envelope of the maskee H ′ (t) can be preserved even if it becomes the masker H (t). As a result, since the spectrum and intonation of the masker H (t) are almost the same as those of the maskee H '(t), the sense of incongruity is naturally received by the listener.

  (3) When the consonant part is extracted by the partial extraction unit 30, the masker H (t) replaces only the consonant part with respect to the masky H '(t), or replaces the consonant part and then silences the vowel part. Generated by substituting or processing. Therefore, it is possible to suppress the increase in the volume (sound pressure level) of the masker H (t) and hence the indoor noise level as much as possible.

  (4) No masker H (t) is output when there is no maskee H '(t) on the time axis, that is, when there is no conversation. That is, both overlap substantially in time. Therefore, an increase in the room noise level due to the masker H (t) during “no sound” when no sound is generated can be suppressed.

  (5) Dissipating a feeling of discomfort due to intermittent maskers or level fluctuations (disrupted when the conversation is stopped to reduced level) that may occur when using conventional technology, or other sounds (noise / music) that are not related to conversation This reduces the sense of discomfort for speakers, conversers, and other people in the room.

  (6) With respect to the physical sound insulation and private room formation in the prior art, no spatial blockage or movement is required, so that a sense of openness and communication are less likely to be hindered.

  (7) Since the SD controller unit SD and the speaker SP are incorporated in the IT partition 4, the installation and installation of the system can be greatly simplified. In some cases, the microphone Mic may be incorporated in the IT partition 4. In this case, it is further simplified.

  (8) The IT partition 4 itself is subjected to sound absorption processing. Therefore, sound leakage to the adjacent booth can be reduced while increasing the clarity of the conversation voice in the booth.

  (9) The masker H (t) becomes a signal whose electrical signal correlation is not so high as that of the maskee H ′ (t) (original voice) by processing such as substitution, deletion, inversion, and rotation. Therefore, abnormalities due to feedback such as howling are less likely to occur during the operation of the speech information confidential system 100.

  (10) In the real time mode of the SD controller unit SD according to the present embodiment, time reversal or time rotation is performed on the signal of the change target portion. When time reversal is performed, it is possible to generate a masker H (t) effective for information disturbance while preserving the signal envelope. However, in the case of time reversal, there may be a case where a audible difference does not occur between the maskee H ′ (t) and the masker H (t). On the other hand, when time rotation is performed, it has been found by experiments by the present inventor that the auditory impression of the maskee H ′ (t) and the masker H (t) changes slightly.

  For information concealment / hearing, it is a problem that the maskee H '(t) and the masker H (t) are too auditoryly similar, but it is also a problem that they are too different. This is because the auditory sense has a property of distinguishing and recognizing different properties. Therefore, in the case of the time rotation described above, a masker H (t) that is not too close to the auditory sense and not too far away and is just good for information hiding can be provided.

  (11) In the partial extraction unit 30, when approximately one mountain signal is extracted as the signal of the change target part, cutting and pasting are performed at the part where the signal level of the maskee H '(t) is low, so time Click noise due to inversion / rotation processing is reduced. That is, if the masker H ′ (t) is continuous in time, the masker H (t) is also substantially continuous. It is difficult for the envelope shape to collapse (intonation collapse) due to the windowing process.

  (12) When the partial extraction unit 30 extracts a substantially one-crested signal as a signal to be changed, and when the extracted signal is subjected to time rotation processing, the shape of the masker spectrum or envelope Are almost conserved and similar to those of Muskie. Therefore, it is possible to effectively perform sound field information disturbance (concealment of audio content) while suppressing an increase in indoor noise level and click noise to a minimum.

  Heretofore, the configuration and operation of the audio information secret system 100 according to the embodiment and the SD controller unit SD included therein have been described. This embodiment is an exemplification, and it is understood by those skilled in the art that various modifications can be made to each component and combination of processes, and such modifications are within the scope of the present invention.

  In the embodiment, the case where the masker H (t) is output from one side of the adjacent booth has been described, but the present invention is not limited to this. For example, the masker H (t) may be output from the left and right sides of the adjacent booth by signal addition. FIG. 12 is a block diagram schematically showing the function and configuration of the speech information secret system according to the first modification. The audio information secret system according to the first modification includes a microphone Mic, an SD controller unit SD, four speakers SPa to SPd (SPd is not shown), and four power amplifiers PAa to PAd (PAd is not shown). Four adders 210a to 210d (210d not shown).

The audio signal that has undergone the processing in the SD controller unit SD is the adder 210a corresponding to the left speaker SPa of the booth 2, the adder 210b corresponding to the right speaker SPb of the booth 2, and the adjacent booth adjacent to the left of the booth 2. The signal is input to the adder 210c corresponding to the left speaker SPc of 2 ′ and the adder 210d (not shown) corresponding to the right speaker SPd (not shown) of the adjacent booth adjacent to the right of the booth 2. The audio signals input to the adders 210a to 210d are output from the speakers SPa to SPd via the corresponding power amplifiers PAa to PAd. The adder acquires and adds the audio signal that has undergone processing in the SD controller unit SD from the booths adjacent to the booth to which the speaker to which the speaker is connected outputs audio.
According to this modification, since the masker H (t) is output from both the left and right sides of the adjacent booth 2 ′, the conversation contents in the booth 2 are difficult to be transmitted to the listener 8.

Further, a PNC (Passive Noise Controller) may be used in combination to reduce the level of the maskee H ′ (t). The PNC intends to use a known ANC (Active Noise Control) adaptively at the time of adjustment, and to fix and use the set parameters at the time of operation.
FIG. 13 is a block diagram schematically showing the function and configuration of the audio information secret system according to the second modification. In this modification, the SD controller unit SD in FIG. 12 is replaced with a part surrounded by a broken line in FIG. In this part, the SD controller unit SD and the PNC unit PNC are provided in parallel, and an audio signal from the microphone Mic is input to the SD controller unit SD and the PNC unit PNC. A switch SW1 is provided on the output side of the SD controller unit SD, and the operation of the SD controller unit SD is controlled by the switch SW1. The output of the switch SW1 and the output of the PNC unit PNC are added by an adder 406 and output as sound from the speaker SP via the power amplifier PA.

  In this modification, the head torso simulator HATS (HATS: Head and Torso Simulator) connected to the sound source 402 via the amplifier 404 is placed at the speaker position P, and the PNC unit PNC is identified. The switch SW1 is opened to turn off the operation of the SD controller unit SD, and an appropriate audio signal is emitted from the HATS so that the output of the microphone Mic ′ placed at the listener position Q in the adjacent booth 2 ′ is minimized. System identification is performed by adaptively operating.

At this time, the impulse response including the microphone Mic and the speaker SP is −h (x), and the absolute value is substantially equal to that h (x) between the PNC speaker and the listener. Thereafter, the switch SW1 is closed, and the PNC unit is operated with the identified parameters fixed. Then, since the positions P and Q of the speaker and the listener and the positions of the microphone Mic and the speaker SP are substantially fixed, the level of the maskee H ′ (t) is effectively reduced, and the masker H (t) is dominant. Become. As a result, the effect of information masking is enhanced. If the level of the masker H (t) is lowered as necessary, the level of the entire system including the maskee H ′ (t), that is, the noise level in the room can be further reduced.
Note that the PNC function described above may be incorporated in a computer in which the SD controller unit SD is incorporated.

  Although ANC / PNC is an existing technology, it is not suitable for controlling a wide sound field all over three dimensions. On the other hand, in the case where the listener's head is present at a substantially fixed position in a narrow space surrounded by the partition of the counter, the sound reduction means is effective even in three dimensions.

  In replacement or deletion of a change target portion such as a consonant portion in the embodiment, a time window such as a Hanning window or zero cross detection may be used together to remove a click sound that may occur at the time of clipping. In this case, the uncomfortable feeling that can be given to the listener 8 or the people in the room is further reduced.

Although the present invention has been described based on the embodiments, the embodiments merely show the principle and application of the present invention, and the embodiments are defined in the claims. Needless to say, many modifications and arrangements can be made without departing from the spirit of the present invention.
For example, it is an example of a technique in which a plurality of processed sounds are radiated on the original sound.

  2 booths, 4 IT partitions, 6 customers, 8 listeners, 10 storage devices, 20 A / D units, 30 partial extraction units, 72 output units, 90 partial change units, 100 voice information secrecy systems, SD SD controller units, SPs Speaker, Mic microphone.

Claims (7)

A partial extraction unit that extracts a signal of a change target portion based on a waveform of the voice signal from a voice signal representing the voice being uttered;
A partial changing unit that changes the signal of the change target portion extracted by the partial extracting unit;
An output unit configured to output a signal of the change target portion changed by the partial change unit to an audio output unit capable of outputting audio to a region where the voice being spoken is received; Change device.   The partial extraction unit is a signal in a section sandwiched between a first time before the peak of the envelope of the waveform of the audio signal and a second time after the peak, and a substantially one mountain-shaped signal, The sound changing device according to claim 1, wherein the sound changing device is determined as a signal of a change target portion.   The voice changing device according to claim 1, wherein the partial extraction unit extracts a signal of a consonant part as a signal of a change target part.   The voice change device according to any one of claims 1 to 3, wherein the partial change unit rotates a waveform along a time axis of a signal of the change target portion extracted by the partial extraction unit.   The apparatus further comprises a timing adjustment unit that adjusts the timing at which the signal of the change target portion changed by the partial change unit is output from the output unit according to the time taken for propagation of the voice during the utterance. The voice changing device according to any one of claims 1 to 4. Sound collecting means for receiving the voice being uttered and generating a voice signal representing the voice;
A sound changing device for changing a sound signal generated by the sound collecting means;
Voice output means for converting the voice signal changed by the voice changing device into voice and outputting the voice to the area where the voice being spoken is received,
The voice changing device is
A partial extraction unit that extracts a signal of a change target portion based on a waveform of the audio signal from the audio signal generated by the sound collecting unit;
A partial changing unit that changes the signal of the change target portion extracted by the partial extracting unit;
And an output unit that outputs the signal of the change target portion changed by the partial change unit to the voice output unit. Extracting a signal to be changed from a voice signal representing a voice being uttered based on a waveform of the voice signal;
Changing the extracted signal of the change target portion;
Converting the changed signal of the change target portion into a voice, and outputting the converted voice to a region where the voice being spoken is listened to.

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