JP2008288910A - Sound pickup device - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To reduce noise by using directivity when sound is picked up in a sound pickup device and specifying a noise generation period. <P>SOLUTION: A directivity generating portion 330 generates a directivity signal having directivity to a surrounding specific direction based on a voice signal from a voice input portion 310 supplied through an amplifier 320. When detecting noise from the directivity signal supplied from the directivity generating portion 330, a noise detecting portion 360 generates a signal indicating a noise removal period according to a noise generation period. According to the noise removal period supplied from the noise detecting portion 360, a noise reducing processor 370 removes noise in the directivity signal supplied from the directivity generating portion 330 in the noise removal period, and does not remove the noise in the noise removal period. <P>COPYRIGHT: (C)2009,JPO&INPIT

Description

  The present invention relates to a sound collection device, and more particularly to a sound collection device that reduces noise in an audio signal.

  An imaging device such as a video camera often incorporates a microphone in order to pick up sound sources around the imaging device in accordance with the imaging of the subject. The main purpose of this microphone is to pick up sound that matches the angle of view of the lens, but in recent years, three-dimensional surround sound pickup has also been performed on the entire surroundings.

  In addition to the sound collection mechanism, such an imaging apparatus includes a disk drive mechanism such as a DVD (Digital Versatile Disc) or HDD (Hard Disc Drive) as a recording medium, autofocus, power zoom, optical image stabilization, and the like. A lens drive mechanism is also installed. In addition, a liquid crystal monitor opening / closing mechanism operated by a user, various operation switches, and the like are also included. The operating sound of these mechanisms is incident as noise on the above-described sound collecting mechanism, and according to the sound collecting mechanism, it is a factor that lowers the S / N ratio (Signal to Noise ratio) of the target sound collecting sound. Become. Although these noises should be suppressed on the generation side, they are becoming increasingly difficult due to the downsizing and high functionality of the imaging device.

  In addition, these noises are caused by vibrations that propagate through the cabinet and acoustic noises that propagate as sound in the air at the same time, which complicates the noise transmission path to the microphone. Therefore, by taking a structure that floats the microphone from the cabinet with an insulator such as a rubber damper or a structure that floats in the air with a rubber wire or the like, the vibration transmitted from the cabinet is absorbed and noise is not transmitted. In such a passive method alone, a sufficient noise reduction effect has not been obtained.

  Furthermore, such noise is often generated instantaneously in a short timing period, for example, several milliseconds to several hundred milliseconds, as represented by a click sound, and is reduced by noise reduction techniques such as an adaptive filter. There were many cases where was not in time.

On the other hand, a technique for reducing noise by using a masking effect in human hearing has been proposed. For example, an apparatus for reducing noise by switching sound signals by detecting noise generation timing from the outside has been proposed (for example, see Patent Document 1).
Japanese Patent Laying-Open No. 2005-303681 (FIG. 1)

  The above-described conventional technology removes the above-described shock noise, touch noise, click noise, and the like so as not to be recognized by human hearing, and is effective when the noise generation period can be specified.

  However, in the prior art, when noise other than noise is mixed in the input signal, or when the noise timing cannot be obtained from the driving device, the noise generation period cannot be specified, and therefore noise cannot be removed. There was a problem.

  The present invention has been made in view of such a situation, and an object thereof is to specify a noise generation period using directivity at the time of sound collection and reduce noise.

  The present invention has been made to solve the above-described problems, and a first aspect of the present invention is that a first direction is based on a plurality of surrounding sound signals and sound input means for inputting a plurality of surrounding sound signals. Directivity generating means for generating a first directivity signal having directivity and a second directivity signal having directivity in the second direction, and noise for removing a noise band from the first directivity signal Removing means, noise recognizing means for recognizing noise included in the second directivity signal, noise removing period generating means for generating a signal indicating a noise removing period according to the recognized noise generation period, When the noise removal period is indicated, the output of the noise removal means is selected, and when the noise removal period is not indicated, the first directivity signal is selected. With selection means A sound pickup device, characterized by Bei. Accordingly, there is an effect that the presence or absence of noise removal from the first directional signal is selected according to the generation period of the noise included in the second directional signal.

  In the first aspect, the voice input unit may include a plurality of bidirectional microphones and one omnidirectional microphone. The voice input means may include a plurality of omnidirectional microphones. The voice input means may include a plurality of unidirectional microphones and a single bidirectional microphone. Based on these, first and second directivity signals are generated in the directivity generating means.

  The first aspect may further include rotation coefficient generation means for generating a rotation coefficient indicating a predetermined direction, and the directivity generation means may be configured so that the direction indicated by the rotation coefficient is the first direction. The first directivity signal may be generated, and the second directivity signal may be generated if the direction indicated by the rotation coefficient is the second direction. Thereby, the first and second directivity signals are generated with the rotation coefficient as a reference.

  In the first aspect, the noise recognizing means evaluates an output from a convolution operation of a wavelet signal having a mean value of zero in a predetermined period approximated to the noise and a second directivity signal. The above noise recognition may be performed. This brings about the effect that the presence or absence of noise removal is selected according to the noise recognition result in the time domain.

  In the first aspect, the noise recognizing unit performs the noise recognition using an evaluation value as a correlation between a pattern signal approximated to the frequency spectrum of the noise and the second directivity signal subjected to Fourier transform. You may do it. This brings about the effect that the presence or absence of noise removal is selected according to the noise recognition result in the frequency domain.

  In the first aspect, the noise removing unit can be realized by a filter that removes a noise band. In this case, the noise removal unit may adaptively change the removal band and the pass band of the filter based on the frequency of the noise recognized by the noise recognition unit.

  In the first aspect, the selection means may be realized by a cross fade switch. This brings about the effect of crossfading with a predetermined time constant when switching the presence or absence of noise removal.

  According to a second aspect of the present invention, there is provided audio input means for inputting a plurality of surrounding audio signals, a first directional signal having directivity in a first direction based on the plurality of audio signals, and a first Directivity generating means for generating a second directivity signal having directivity in two directions, noise removing means for removing a noise band from the first directivity signal, and the signal from which the noise band has been removed A signal interpolation unit for performing interpolation, a noise recognition unit for recognizing noise included in the second directional signal, and a noise removal period for generating a signal indicating a noise removal period according to the recognized noise generation period When the generation means and the noise removal period are indicated, the output of the signal interpolation means is selected, and when the noise removal period is not indicated, the first directivity is selected. Select signal It may be a sound pickup apparatus characterized by comprising selecting means for. As a result, the presence or absence of noise removal from the first directional signal is selected according to the generation period of the noise included in the second directional signal, and the first directional signal from which noise has been removed is interpolated. It brings about the effect of improving the audible masking effect.

  In the second aspect, the signal interpolation means includes an interpolation source signal generation means for generating an interpolation source signal for the interpolation, and an extra-interpolation removal means for removing an area other than the noise band from the interpolation source signal. Level envelope generating means for generating a level envelope of the first directional signal, level coefficient generating means for generating a level coefficient for the interpolation based on the level envelope, and based on the level coefficient Level modulation means for modulating the output of the non-interpolation removal means, and synthesis means for synthesizing the output of the noise removal means and the output of the level modulation means and outputting them to the selection means. In this case, the level modulation means may further modulate the output of the non-interpolation removal means based on a level masked on human hearing. Further, the interpolation source signal generation means is a predetermined signal of a plurality or a single periodic signal having a predetermined waveform and a predetermined period, a white noise signal having a uniform level in a voice band, or a predetermined period of the periodic signal and the white noise signal. Any one of the mixed signals based on the mixing ratio may be generated.

  In the second aspect, the signal interpolation means includes an interpolation source signal generation means for generating an interpolation source signal for the interpolation, and an extra-interpolation removal means for removing an area other than the noise band from the interpolation source signal. A spectrum envelope generating means for generating a frequency spectrum envelope of the output of the noise removing means; a spectrum coefficient generating means for generating a spectrum coefficient for the interpolation based on the spectrum envelope; and based on the spectrum coefficient. Spectrum modulating means for modulating the output of the non-interpolation removing means, level envelope generating means for generating a level envelope of the first directional signal, and a level for the interpolation based on the level envelope Level coefficient generating means for generating coefficients, and level modulating means for modulating the output of the spectrum modulating means based on the level coefficients May be synthesized with the outputs of the above-level modulation means for said noise removing means comprises a synthesizing means for outputting to said selection means. In this case, the noise removal unit and the non-interpolation removal unit may be realized by a filter that adaptively changes the removal band and the pass band based on the frequency of the noise recognized by the noise recognition unit.

  According to the present invention, it is possible to obtain an excellent effect that noise can be reduced by specifying a noise generation period using directivity during sound collection.

  Next, embodiments of the present invention will be described in detail with reference to the drawings.

  FIG. 1 is a diagram illustrating a configuration example of a sound collection device 300 according to an embodiment of the present invention. The sound collection device 300 includes an audio input unit 310, an amplifier 320, a directivity generation unit 330, a timing generation unit 340, a rotation coefficient generation unit 350, a noise detection unit 360, a noise reduction processing unit 370, An encoding processing unit 380 and a recording / reproducing unit 390 are provided.

  The voice input unit 310 acquires and inputs a surrounding voice signal, and is realized by, for example, a plurality of microphones. The amplifier 320 amplifies the audio signal from the audio input unit 310 and supplies it to the directivity generation unit 330.

  The directivity generation unit 330 generates a directivity signal having directivity in a specific direction around the sound signal from the sound input unit 310 supplied via the amplifier 320. The direction having directivity follows the rotation coefficient supplied from the rotation coefficient generation unit 350. As a result, a directional signal having directivity in each of a plurality of directions is obtained. Of these, a directional signal used for noise recognition is supplied to the noise detection unit 360 via the signal line 338, A directivity signal used as an original audio signal other than the above is supplied to the noise reduction processing unit 370 via the signal line 339.

  The timing generation unit 340 generates operation timings in the directivity generation unit 330, the rotation coefficient generation unit 350, the noise detection unit 360, and the noise reduction processing unit 370. The timing generator 340 divides one audio sampling period “1 / Fs” into m for upsampling processing described later, and generates a timing signal for each sampling period “1 / (m · Fs)”. That is, the sampling frequency is “m · Fs”. The timing signal generated by the timing generation unit 340 is supplied to each unit via a signal line 349.

  The rotation coefficient generation unit 350 generates a rotation coefficient indicating the directivity direction of the directivity signal generated by the directivity generation unit 330. The rotation coefficient generated by the rotation coefficient generation unit 350 is supplied to the directivity generation unit 330 via the signal line 359.

  The noise detection unit 360 detects noise from the directivity signal supplied from the directivity generation unit 330 via the signal line 338. When noise is detected, the noise detection unit 360 generates a signal indicating a noise removal period according to the noise generation period. The noise removal period generated by the noise detection unit 360 is supplied to the noise reduction processing unit 370 via the signal line 369.

  The noise reduction processing unit 370 removes noise included in the directivity signal supplied from the directivity generation unit 330 according to the noise removal period supplied from the noise detection unit 360. The directivity signal that has been subjected to noise removal processing by the noise reduction processing unit 370 is supplied to the encoding processing unit 380 via signal lines 371 to 376. These signal lines 371 to 376 correspond to 5.1 channel surround signals described later.

  The encoding processing unit 380 performs an encoding process on each signal supplied from the noise reduction processing unit 370. The recording stream signal encoded by the encoding processing unit 380 is supplied to the recording / reproducing unit 390 via the signal line 389.

  The recording / reproducing unit 390 records or reproduces the recording stream signal supplied from the encoding processing unit 380 on a recording medium. The recording / playback unit 390 may record the video signal together with the audio signal input from the audio input unit 310, but the description thereof is omitted in the embodiment of the present invention.

  FIG. 2 is a diagram showing the arrangement and directivity characteristics of 5.1 surround sound sources. The 5.1 channel of the surround sound source includes a directivity pattern 591 in the front direction (FRT: Front), a directivity pattern 592 in the front left direction (FL: Front Left), and a front right direction (FR: Front) with reference to the sound collection device 300. Right channel pattern 593, rear left direction (RL: Rear Left) direction pattern 594 and rear right direction (RR: Rear Right) directivity pattern 595, and low frequency band of all directivity direction pattern 596 (LF: Low Frequency) 0.1 channel. The 0.1 channel in the low frequency band is for obtaining a low-weight sound feeling of about 100 Hz or less.

  A surround sound field can be obtained by picking up and recording such a surround sound source and playing it back on an existing surround compatible system. The sound collection and sound source creation of the surround sound field described above is left to the production intention and know-how of the producer, but it is conscious of the 5.1 channel surround sound field reproduction standard ITU (International Telecommunication Union) -R standard. Is often done. In this standard, the front speaker direction (FRT) is 0 degrees, the front left direction (FL) is 30 degrees, the front right direction (FR) is 30 degrees, and the rear left direction (RL) is 100 to 120 as the playback speaker arrangement. It is recommended that the rear right direction (RR) be 100 to 120 degrees.

  FIG. 3 is a diagram showing an example of vector amount extraction in the embodiment of the present invention. In the embodiment of the present invention, a vector as a sound source and a vector for noise detection are set for each direction around the sound collection device.

  The FRT vector 631 is a vector for the front direction, the FL vector 632 is a vector for the front left direction, the FR vector 633 is a vector for the front right direction, the RL vector 634 is a vector for the rear left direction, and the RR vector 635. Is a vector for the backward right direction. Note that the 0.1 channel in the low frequency band has a long wavelength, has little directivity, and is considered to be only a size, so it is treated as a scalar quantity.

  Noise vectors A and F are vectors assuming noise generated from the lens driving mechanism, noise vector B is a vector assuming noise generated from the grip part (gripping part), and noise vector C is assumed noise generated from the disk mechanism. The noise vector D is a vector assuming noise generated from an LCD (Liquid Crystal Display) monitor, and the noise vector E is a vector assuming noise generated from various operation switches.

  As described above, by extracting the vector amount in which the direction and the magnitude (sound collection level) are combined with the noise incident from each direction, only the noise can be easily recognized. In addition, when the sound source direction and the noise direction match, the vector amount in the sound source direction is also used for noise recognition. The sound collection image at this time is a solid line 620 surrounding each vector in FIG.

  FIG. 4 is a diagram illustrating an example of a polar pattern by the sound collection device according to the embodiment of the present invention. The polar pattern is a polar coordinate display of the sensitivity level from the entire circumference of each microphone in the sound collection device. In this figure, the front direction is 0 degree, the sensitivity level in the radial direction is relative, and the center is the sensitivity zero point.

  FIG. 4A shows an omnidirectional (omnidirectional) polar pattern having the same level of sensitivity characteristics in all directions. FIG. 4B is a polar pattern of primary (single) directivity, and is used when directivity is given in a single direction. In this example, it has directivity in the 0 degree direction. FIG. 4C shows a polar pattern with a secondary directivity having a direction selectivity stronger than the primary directivity. FIGS. 4D and 4E are called bi-directionality, and have a maximum sensitivity in a direction opposite to the certain direction and the direction, and zero sensitivity in the direction of 90 degrees with them. 4D and 4E have characteristics orthogonal to each other. Further, the positive electrode (+) characteristic and the negative electrode (−) characteristic are opposite to each other, and the signal phase of both is shifted by 180 degrees. These directivity characteristics can be generated by a single microphone or a combination calculation of a small number of microphones.

  In the embodiment of the present invention, these microphones are mounted in the sound collection device as the sound input unit 310 or are externally mounted. These microphones simultaneously collect sound and noise from a plurality of directions.

  FIG. 5 is a diagram showing a first arrangement example of microphones in the embodiment of the present invention. In this first arrangement example, an omnidirectional microphone 411 and bidirectional microphones 412 and 413 are arranged.

  The omnidirectional microphone 411 is a microphone having no directivity. As shown in FIG. 4D, the bidirectional microphone 412 is a microphone having directivity in both the right direction and the left direction, and is disposed in the front direction relative to the omnidirectional microphone 411. As shown in FIG. 4E, the bidirectional microphone 413 is a microphone having directivity in both the front direction and the backward direction, and is disposed in the backward direction relative to the omnidirectional microphone 411. The positional relationship between the microphones is an example and is not limited to this. For example, the microphones may be arranged three-dimensionally.

  FIG. 6 is a diagram showing a synthesis example of sound sources of microphones according to the first arrangement example according to the embodiment of the present invention. This sound source synthesis mechanism is included in the directivity generation unit 330 and includes level variable units 422 and 423 and an addition synthesis unit 426.

  The level variable unit 422 multiplies the lateral bidirectional signal supplied from the bidirectional microphone 412 by Ks. The level varying unit 423 multiplies the vertical bidirectional signal supplied from the bidirectional microphone 413 by Kc. Here, Ks and Kc are rotation coefficients determined by the directivity direction. This rotation coefficient will be described later.

  The adder / synthesizer 426 performs an averaging process on the three signals of the omnidirectional signal supplied from the omnidirectional microphone 411, the signal supplied from the level variable unit 422, and the signal supplied from the level variable unit 423. To be synthesized. The sound source synthesized by the adding and synthesizing unit 426 becomes a sound source having arbitrary directivity.

Here, the lateral bi-directional signal (FIG. 4D) is a sine function sin (t) at time t, and the vertical bi-directional signal (FIG. 4E) is a cosine function cos (t) at time t. When expressed as t), the sound source X synthesized by the addition synthesis unit 426 can be represented by the following equation. In this equation, “1” corresponds to an omnidirectional signal (FIG. 4A).
X = (1 + Ks · sin (t) + Kc · cos (t)) / 2

  FIG. 7 is a diagram showing a rotation coefficient in the embodiment of the present invention.

  The rotation coefficient Ks611 draws a sine curve according to the directivity rotation angle φ, and the rotation coefficient Kc612 draws a cosine curve according to the directivity rotation angle φ. That is, the rotation coefficients Ks611 and Kc612 are real numbers corresponding to the directivity rotation angle φ in the range from −1 to 1.

  When the rotation angle φ is 0 degree, Ks = 0 and Kc = 1, and only the bidirectional signal from the bidirectional microphone 413 is input to the adder / synthesizer 426. When the rotation angle φ is 45 degrees, both Ks and Kc are reciprocals of the square root of 2 (≈0.7), and the bidirectional signals from the bidirectional microphones 412 and 413 are added and combined at the same level. At 426, the averaging process is performed, and the omnidirectional signal is further subjected to an averaging process. This is shown in FIG.

  That is, when the rotation angle φ is 45 degrees, the two bi-directional signals are subjected to addition averaging processing at the same level, so that the reverse phase portion by the broken line is canceled and the in-phase portion by the solid line remains, and FIG. A signal of directivity 511 at is obtained. Then, the signal of directivity 511 and the signal of non-directivity 512 are added and averaged to cancel the reverse phase portion indicated by the broken line, the in-phase portion indicated by the solid line remains, and the rotation angle φ in FIG. A 45 degree directivity 513 signal is obtained.

  Similarly, when the rotation angle φ is 90 degrees, only the bidirectional signal from the bidirectional microphone 412 is input to the adder / synthesizer 426. When the rotation angle φ is 90 to 180 degrees, the positive and negative polarities of the bidirectional signal from the bidirectional microphone 413 are inverted and synthesized by multiplying Kc by a negative coefficient. When the rotation angle φ is 180 to 270 degrees, the positive and negative polarities of the bidirectional signals from the bidirectional microphones 412 and 413 are inverted and synthesized by multiplying Ks and Kc by a negative coefficient. When the rotation angle φ is 270 to 0 degrees, the positive and negative polarities of the bi-directional signal from the bi-directional microphone 412 are inverted and synthesized by multiplying Ks by a negative coefficient.

  In this way, by setting the rotation coefficients Ks and Kc, a signal having directivity at an arbitrary rotation angle φ can be generated. A surround sound source can be generated by using the signal generated in this way. Then, by distributing these sound sources into the vectors shown in FIG. 3, it is possible to distinguish the noise detection signal from the original audio signal.

  FIG. 9 is a diagram illustrating a second arrangement example of the microphones according to the embodiment of the present invention. In the second arrangement example, four microphones omnidirectional microphones 431 to 434 are arranged. The distance between these omnidirectional microphones 431 to 434 is, for example, about 10 to 15 millimeters. Further, the straight line connecting the omnidirectional microphones 431 and 433 and the straight line connecting the omnidirectional microphones 434 and 432 may be orthogonal to each other, and the mutual positional relationship is not limited to this.

  None of these omnidirectional microphones 431 to 434 have directivity in a specific direction, but a signal having directivity in an arbitrary direction can be generated by combining them.

  FIG. 10 is a diagram illustrating an example of generation of directivity characteristics of microphones in the second arrangement example according to the embodiment of the present invention.

  When the frequency characteristic is adjusted by subtracting the sound source of the omnidirectional microphone 433 from the sound source of the omnidirectional microphone 431, a bidirectional signal 506 as shown in FIG. Further, when the frequency characteristics are adjusted by subtracting the sound source of the omnidirectional microphone 432 from the sound source of the omnidirectional microphone 434, a bidirectional signal 507 as shown in FIG. 10B is generated. Furthermore, an omnidirectional signal is generated by adding any combination of the sound sources of the omnidirectional microphones 431 to 434.

  FIG. 11 is a diagram showing a synthesis example of the sound sources of the microphones in the second arrangement example according to the embodiment of the present invention. This sound source synthesis mechanism is included in the directivity generation unit 330, and includes an addition unit 441, subtraction units 442 and 443, level variable units 444 and 445, and an addition synthesis unit 446.

  The adder 441 generates an omnidirectional signal by performing an averaging process on all the sound sources of the omnidirectional microphones 431 to 434. The subtracting unit 442 subtracts the sound source of the omnidirectional microphone 432 from the sound source of the omnidirectional microphone 434 to generate the lateral bidirectional signal 507 in FIG. The subtracting unit 443 subtracts the sound source of the omnidirectional microphone 433 from the sound source of the omnidirectional microphone 431 to generate the vertical direction bi-directional signal 506 in FIG.

  The level variable unit 444 multiplies the horizontal bidirectional signal 506 in FIG. 10B by Ks. The level variable unit 445 multiplies the vertical bidirectional signal 507 in FIG. 10A by Kc. These direction coefficients Kc and Ks are the same as those described with reference to FIG.

  The adder / synthesizer 446 synthesizes the three signals of the omnidirectional signal supplied from the adder 441, the signal supplied from the level variable unit 444, and the signal supplied from the level variable unit 445 through an addition averaging process. Is. The sound source synthesized by the adding and synthesizing unit 446 becomes a sound source having arbitrary directivity.

  In the microphone sound source synthesis example in the second arrangement example, the output of the adder 441 is equivalent to the output of the omnidirectional microphone 411 in the first arrangement example, and the output of the subtractor 442 is the first arrangement example. The output of the subtracting unit 443 is equivalent to the output of the bidirectional microphone 413 in the first arrangement example. Therefore, the output from the addition synthesis unit 446 is equivalent to the output from the addition synthesis unit 426.

  FIG. 12 is a diagram showing a third arrangement example of microphones according to the embodiment of the present invention. In this third arrangement example, three microphones, unidirectional microphones 452 and 453 and a bidirectional microphone 451, are arranged. The positional relationship between the microphones is an example and is not limited to this. For example, the microphones may be arranged three-dimensionally.

  The bi-directional microphone 451 is a microphone having directivity in both the right direction and the left direction as shown in FIG. The unidirectional microphone 452 is a microphone having directivity in the front direction as shown in FIG. The unidirectional microphone 453 is a microphone having directivity on the rear side, contrary to FIG. The distance between these microphones is, for example, about 10 to 15 millimeters.

  FIG. 13 is a diagram showing a synthesis example of the sound sources of the microphones in the third arrangement example according to the embodiment of the present invention. This sound source synthesis mechanism is included in the directivity generation unit 330 and includes level variable units 461 to 463 and an addition synthesis unit 466.

  The level variable unit 461 multiplies the sound source of the bidirectional microphone 451 by Ks. The level variable unit 462 multiplies the sound source of the unidirectional microphone 452 by (1 + Kc). The level variable unit 463 multiplies the sound source of the unidirectional microphone 453 by (1−Kc). These direction coefficients Kc and Ks are the same as those described with reference to FIG.

  The adder / synthesizer 466 synthesizes the three signals supplied from the level variable units 461 to 463 by the addition averaging process. The sound source synthesized by the adding and synthesizing unit 466 becomes a sound source having arbitrary directivity.

Here, when the vertical bidirectional signal is expressed as a cosine function cos (t) at time t, the sound source of the unidirectional microphone 452 is (1 + cos (t)). The sound source of the unidirectional microphone 453 is (1-cos (t)). When the lateral bi-directional signal is represented as a sine function sin (t) at time t, the sound source Y synthesized by the addition synthesis unit 466 can be represented by the following equation.
Y = ((1 + Kc). (1 + cos (t)) / 2
+ (1-Kc). (1-cos (t)) / 2
+ Ks · sin (t)) / 2

  FIG. 14 is a diagram illustrating a generation example of the directivity characteristics of the microphone in the third arrangement example according to the embodiment of the present invention.

  14A shows the directivity 521 of the sound source of the unidirectional microphone 452, the directivity 522 of the sound source of the unidirectional microphone 453, and the directivity 523 of the sound source of the bi-directional microphone 451. FIG. ing.

  In the expression of the sound source Y synthesized by the adding and synthesizing unit 466, when the rotation angle φ is set to 0 degree, Ks = 0 and Kc = 1. Therefore, the adding and synthesizing unit 466 receives the unidirectional microphone 452. A sound source is output. When the directivity rotation angle φ is set to 45 degrees, both Ks and Kc are reciprocals of the square root of 2 (≈0.7), and therefore, the addition / synthesis unit 466 performs addition averaging processing, and FIG. A unidirectional signal is generated in the 45-degree direction like the directivity 524 in (b).

  Further, when the directivity rotation angle φ is set to 90 degrees, Ks = 1 and Kc = 0, so that a non-directional signal is generated by the level variable units 462 and 463, and the adder / synthesizer 466 is bi-directional. The unidirectional signal is generated in the 90-degree direction by performing an averaging process with the bidirectional signal of the directional microphone 451.

  Similarly, Kc is synthesized with a negative coefficient when the directivity rotation angle φ is in the range of 90 to 180 degrees, and Ks and kc are synthesized with a negative coefficient when the directivity rotation angle φ is in the range of 180 to 270 degrees. Ks is synthesized with a negative coefficient in the range where the rotation angle φ of the sex is 270 to 0 degrees. For example, when the directivity rotation angle φ is set to 315 degrees, Kc is the inverse of the square root of 2 (≈0.7), and Ks is the negative of the inverse of the square root of 2 (≈−0.7). Become. As a result, the addition and synthesis unit 466 performs addition averaging processing, and a unidirectional signal is generated in the direction of 315 degrees as in the directivity 525 of FIG.

In the first to third arrangement examples described above, the technique for obtaining the unidirectional signal as shown in FIG. 4B has been described. However, the secondary directional signal as shown in FIG. 4C is generated. It is also possible. In this case, the synthesized sound source Z can be expressed by the following equation. In this equation, “1” corresponds to an omnidirectional signal (FIG. 4A), sin (t) corresponds to a lateral bidirectional signal (FIG. 4D), and cos ( t) corresponds to the vertical bidirectional signal (FIG. 4E).
Z = ((1 + Ks · sin (t) + Kc · cos (t))
(Ks.sin (t) + Kc.cos (t))) / 2

  According to this secondary directivity signal, since the directivity can be further narrowed, the selectivity of each directivity signal for noise detection described later can be improved.

  The above first to third arrangement examples are examples for explanation, and can be changed within the scope of the object of the present invention as long as the microphones are relatively close to each other. For example, these do not need to be arranged on a straight line or at equal intervals, and in the arrangement example of FIG. 9, three microphones can be similarly configured.

  FIG. 15 is a diagram showing an example of the directivity rotation angle φ in the embodiment of the present invention. The directivity rotation angle φ (651) represents the angle formed by the directivity 650 that rotates clockwise with the front direction as 0 degree.

  According to the above-described sound source synthesizing mechanism, it is possible to synthesize a plurality of directional signals with an arbitrary rotation angle φ around the entire circumference. However, if these directional signals are handled individually, the number of channels to be handled increases, and the processing may become large or complicated. Therefore, in the embodiment of the present invention, each directional signal is treated as a directional stream signal of a single channel or a small number of channels.

  FIG. 16 is a diagram illustrating an example of the contents of a directional stream signal in the embodiment of the present invention. In this figure, the horizontal axis is a direction channel obtained by dividing the entire circumference every 30 degrees as an example. In this example, there are 12 channels ranging from a D_1 channel with a rotation angle φ of 0 degrees, a D_2 channel with a rotation angle φ of 30 degrees, a D_3 channel with a rotation angle φ of 90 degrees, and a D_c channel with a rotation angle φ of 330 degrees. It is shown.

  The vertical axis represents the audio sampling period. When the sampling frequency is Fs, the audio sampling period is “1 / Fs”. In the audio sampling period Ts_0, the directivity signals sampled in order from the D_1 channel are sequentially arranged in the manner of S (01), S (02), and S (03). In the audio sampling period Ts_1, the directional signals sampled in order from the D_1 channel are sequentially arranged in the manner of S (11), S (12), and S (13).

  The signals sampled in this way are sequentially scanned and generated as a single directional stream signal as shown in FIG. This directional stream signal includes vector component levels in both the time axis and direction. That is, the directivity pattern described in the above arrangement example can be regarded as an aggregate of vector quantities having the strongest magnitude in the directivity direction, and by changing the principal axis direction along the rotation angle φ, A vector amount corresponding to the sound collection level in each main axis direction is obtained for each audio sampling period.

  As shown in FIG. 17, when sampling m (m is an integer) direction channels in one audio sampling period “1 / Fs”, the necessary directional stream signal sampling period is “1 / (m · Fs) ". For example, in the example of FIG. 16, since m = 12, the sampling period is “1 / (12 × Fs)”.

  FIG. 18 is a diagram illustrating a configuration example of the directivity generation unit 330 according to the embodiment of the present invention. The directivity generation unit 330 includes an upsampling unit 331, an interpolation filter 332, and a directivity generation unit 333.

  The upsampling unit 331 is for upsampling the audio signal acquired from the audio input unit 310 via the amplifier 320. That is, the audio signal sampled at the sampling frequency Fs is resampled at the sampling frequency “m · Fs” in the upsampling unit 331.

  The interpolation filter 332 removes unnecessary wideband components (false signals) generated by re-sampling in the upsampling unit 331. The interpolation filter 332 is realized by, for example, a low pass filter (LPF).

  The directivity generation unit 333 generates a directivity signal based on the audio signal having the sampling period “1 / (m · Fs)” supplied from the interpolation filter 332. The directivity generation unit 333 generates a directivity signal having directivity corresponding to the rotation coefficient in accordance with the rotation coefficient supplied from the rotation coefficient generation unit 350 via the signal line 359. Here, a directivity signal for noise detection is supplied from the signal line 338 to the noise detection unit 360, and other directivity signals used as original audio signals are supplied from the signal line 339 to the noise reduction processing unit 370. Shall be.

  FIG. 19 is a diagram illustrating a configuration example of the downsampling mechanism according to the embodiment of the present invention. This downsampling mechanism is provided inside the noise reduction processing unit 370 and the noise detection unit 360, and includes a directivity direction extraction unit 371, a decimation filter 372, and a downsampling unit 373.

  The directivity direction extraction unit 371 extracts each directivity signal at a timing synchronized with the sampling frequency “m · Fs” supplied by the signal line 349.

  The decimation filter 372 removes unnecessary aliasing components from the directivity signal extracted by the directivity direction extraction unit 371, and is realized by, for example, a low-pass filter (LPF).

  The down-sampling unit 373 returns the original sampling frequency Fs by multiplying the sampling rate of the directional signal supplied from the decimation filter 372 by “1 / m”.

  Thereby, the noise reduction processing unit 370 corresponds to, for example, the 5.1 channel surround sound source of the front direction, the front left direction, the front right direction, the rear left direction, the rear right direction, and the low frequency band described with reference to FIG. A directional signal can be generated. In addition, the noise detection unit 360 can generate, for example, a directivity signal corresponding to the noise vector described with reference to FIG.

  FIG. 20 is a diagram illustrating a first configuration example of the noise reduction mechanism according to the embodiment of the present invention. In this noise reduction mechanism, a directional signal for noise detection is input via the signal line 118, and other directional signals used as original sound signals are input via the signal line 119. Then, noise reduction processing is performed on this directional signal.

  This noise reduction mechanism includes an interpolation source signal generation unit 130, a noise removal filter 141, an inverse filter 142, a level envelope generation unit 171, a level coefficient generation unit 172, a level modulation unit 173, a synthesis unit 180, A selection switch 190, a noise recognition unit 210, and a noise removal period generation unit 220 are provided. Although it is assumed that the noise recognition unit 210 and the noise removal period generation unit 220 are included in the noise detection unit 360 and other units are included in the noise reduction processing unit 370, the present invention is not limited to this.

  The noise removal filter 141 is a filter that removes a noise band from the directional signal from the audio input unit 310. The noise removal filter 141 is realized by, for example, a BEF (Band Elimination Filter) or the like whose removal target is a single or a plurality of frequency bands. The output of the noise removal filter 141 is supplied to one input of the synthesis unit 180 via the signal line 149.

  The interpolation source signal generation unit 130 generates an interpolation source signal for interpolation. In the embodiment of the present invention, the interpolating signal is synthesized with the directional signal from which the noise band has been removed by the noise removing filter 141, thereby improving the human auditory masking effect. The interpolation source signal generation unit 130 outputs a signal obtained by appropriately mixing a tone signal and a random signal as an interpolation source signal that is a source of the interpolation signal. The configuration of the interpolation source signal generation unit 130 will be described later.

  The inverse filter 142 is a filter that removes other than the noise band from the interpolation source signal generated by the interpolation source signal generation unit 130. The inverse filter 142 has the inverse filter characteristics of the noise removal filter 141, and the stop band of the noise removal filter 141 becomes the pass band of the inverse filter 142, and the pass band of the noise removal filter 141 is the stop band of the inverse filter 142. It becomes. The output of the inverse filter 142 is supplied to the level modulation unit 173 via the signal line 148.

  The level envelope generator 171 continuously detects the level envelope (level envelope) of the directional signal from the audio input unit 310. The output of the level envelope generator 171 is supplied to the level coefficient generator 172 via the signal line 177.

  The level coefficient generation unit 172 generates a level coefficient based on the level envelope supplied from the level envelope generation unit 171. The output of the level coefficient generation unit 172 is supplied to the level modulation unit 173 via the signal line 178.

  The level modulation unit 173 performs level modulation on the interpolation source signal supplied from the inverse filter 142 in accordance with the level coefficient supplied from the level coefficient generation unit 172, and outputs the result as an interpolation signal. The output of the level modulation unit 173 is supplied to one input of the synthesis unit 180 via the signal line 179.

  The synthesizer 180 synthesizes the directivity signal supplied from the noise removal filter 141 via the signal line 149 and the interpolated signal supplied from the level modulator 173 via the signal line 179. The synthesizing unit 180 is realized by an adder, for example. The output of the synthesis unit 180 is supplied to the ON input terminal of the selection switch 190 via the signal line 189.

  The noise recognition unit 210 recognizes noise included in the directivity signal from the voice input unit 310. The output of the noise recognition unit 210 is supplied to the noise removal period generation unit 220 via the signal line 219. When the noise recognition unit 210 recognizes noise, the noise removal period generation unit 220 generates a signal indicating the noise removal period according to the noise generation period. The output of the noise removal period generator 220 is supplied to the control terminal of the selection switch 190 via the signal line 369.

  The selection switch 190 selects the directivity signal supplied from the synthesizing unit 180 via the signal line 189 in the noise removal period according to the signal supplied from the noise removal period generation unit 220 via the signal line 369. If the period is not a noise elimination period, the switch selects a directional signal supplied from the audio input unit 310 via the signal line 119. The output of the selection switch 190 is supplied via the signal line 199 for subsequent processing.

  Although an example of a noise reduction mechanism for one channel of a directional signal is shown here, in actuality, the number of noise reduction mechanisms corresponding to the required number of channels is provided.

  FIG. 21 is a diagram for explaining the masking phenomenon used in the embodiment of the present invention. Human hearing is not aware of the presence of small sounds behind relatively loud sounds, so that human voices are difficult to hear in loud noises. Such a phenomenon is called a masking phenomenon and is known to depend on conditions such as a frequency component, a sound pressure level, and a duration. This auditory masking phenomenon is roughly divided into frequency masking and time masking, and time masking is further divided into simultaneous masking and non-simultaneous masking (continuous masking). This masking phenomenon is applied as high-efficiency encoding that compresses an audio signal to about 1/5 to 1/10 in a CD (compact disc), for example.

  In FIG. 21, the elapsed time is shown in the horizontal direction, and the absolute value of the signal level for each time is shown in the vertical direction. As shown in FIG. 21A, when the signal A is input at a predetermined level and the signal B is input at a predetermined level across a gap period of no signal, the human audibility level is schematically shown in FIG. Indicated. That is, in human hearing, even after the signal A disappears, the pattern of the signal A remains for a while like the region 91 while the sensitivity decreases. Such a phenomenon is called forward (forward) masking, and even if another sound is present during this period, it becomes inaudible for human hearing. Also, just before the signal B is input, the same sensitivity decrease occurs as in the region 92. This is called backward (reverse) masking, and even if another sound is present during this period, it becomes inaudible for human hearing.

  Usually, the forward masking amount is larger than the backward masking amount, and it occurs about several hundreds mS at the maximum although it depends on the conditions in terms of time. Under certain conditions, the gap period shown in FIG. 21A is not perceived by hearing for several milliseconds to several tens of milliseconds, and a phenomenon occurs in which the signals A and B are heard as continuous sounds. Such a phenomenon is described in R.A. Research paper on gap detection in Plomp (1963), research paper by Miura (JAS. Journal 94.November issue), and introduction to auditory psychology (B.C.J. Moore, written by Kengo Ogushi, Seishin Shobo, No. 1) As shown in Chapter 4 / Temporal resolution of auditory system, it is known to have the following characteristics.

(First characteristic): If there is a correlation between the frequency bands of the signals A and B, the gap length becomes large. Further, if the continuity between the signal A and the signal B is maintained in terms of frequency, the gap length increases.
(Second characteristic): The band length of the band signal is larger than that of the single sine wave signal.
(Third characteristic): When the levels of the signals A and B are the same, the gap length increases as the signal level decreases, and the gap length does not change when the signal level increases beyond a certain level.
(4th characteristic): The gap length becomes large when the level of the signal B is smaller than the signal A.
(Fifth characteristic): The gap length increases as the center frequency included in the signal decreases, and the gap length decreases as the center frequency increases.

  In the embodiment of the present invention, the level coefficient generation unit 172 generates a level coefficient for interpolation in consideration of these five characteristics. For example, the level coefficient generation unit 172 increases the gap period when the audio level is low (third characteristic), and increases the gap period when the audio level tends to be lower than when the audio level tends to increase with time. Increase the length (fourth characteristic).

  FIG. 22 is a diagram illustrating a configuration example of the interpolation source signal generation unit 130 in the embodiment of the present invention. The interpolation source signal generation unit 130 includes a tone signal generation unit 131, a white noise signal generation unit 132, and a mixing unit 133.

  The tone signal generator 131 generates a tone signal composed of a single or plural sine waves or pulse waves having a predetermined period. This tone signal has a single or a plurality of peaks at a predetermined frequency in terms of frequency characteristics.

  The white noise signal generator 132 generates a white noise signal (random signal) having a uniform level in the entire audio band. The white noise signal generator 132 is realized by, for example, an M-sequence random number generator.

  The mixing unit 133 outputs a mixed signal obtained by mixing the tone signal generated by the tone signal generating unit 131 and the white noise signal generated by the white noise signal generating unit 132 at a predetermined mixing ratio as an interpolation source signal. The output of the mixing unit 133 is supplied to the inverse filter 142 via the signal line 139.

  The predetermined mixing ratio is appropriately set according to the noise removal band characteristic of the noise removal filter 141. However, either one may be set to zero, and only the tone signal or only the white noise signal may be output as the interpolation source signal.

  FIG. 23 is a diagram illustrating an example of frequency characteristics of the noise removal filter 141 and the inverse filter 142 according to the embodiment of the present invention. Here, the frequency is shown in the horizontal direction, and the pass signal level of the filter is shown in the vertical direction.

  FIG. 23A shows an example of the frequency characteristic of the noise removal filter 141. Here, it is shown that the filter has three of fa, fb, and fc as the center frequencies of the removal band. On the other hand, FIG. 23B shows an example of the frequency characteristic of the inverse filter 142, and shows that the filter has the center frequencies fa, fb, and fc as passbands, contrary to the noise removal filter 141.

  That is, in this example, it can be seen that the center frequencies fa, fb, and fc are treated as noise bands, the noise removal filter 141 treats the noise bands as removal bands, and the inverse filter 142 treats the noise bands as pass bands.

  FIG. 24 is a diagram illustrating a configuration example of the level envelope generation unit 171 according to the embodiment of the present invention. The level envelope generation unit 171 includes an absolute value generation unit 174 and a smoothing unit 175.

  The absolute value generation unit 174 generates an absolute value of the directional signal supplied via the signal line 119. The smoothing unit 175 extracts and smoothes a low frequency component from the directivity signal absolute valued by the absolute value generation unit 174, and is realized by, for example, a low-pass filter (LPF). By this smoothing, it is possible to remove the influence caused by a rapid level change such as instantaneous noise.

  FIG. 25 is a diagram illustrating an example of a process performed by the level envelope generation unit 171 in the embodiment of the present invention. FIG. 25A is a waveform example of a directional signal (audio signal) supplied to the level envelope generation unit 171 via the signal line 119. This directivity signal is converted into an absolute value by the absolute value generation unit 174, thereby forming a waveform as shown in FIG.

  Then, the absolute value-directed directional signal having the waveform of FIG. 25B is smoothed by the smoothing unit 175, thereby forming an envelope like the thick line shown in FIG.

  A level coefficient is generated by the level coefficient generation unit 172 based on the level envelope generated in this manner, and an interpolation signal is generated by controlling the level modulation unit 173 with this level coefficient.

  FIG. 26 is a diagram showing an example of the interpolation signal in the embodiment of the present invention. In this example, the correction signal 21 is generated so as to maintain the frequency continuity between the signal A and the signal B based on the level envelope generated by the level envelope generation unit 171. Thereby, the gap length can be increased by the above-described first characteristic.

  FIG. 27 is a diagram showing another example of the interpolation signal in the embodiment of the present invention. In this example, the correction signal 22 is generated to compensate for the shortage ΔS between the forward masking and backward masking and the signal B shown in FIG. This prevents gaps from being felt in the sense of hearing. That is, in the example of FIG. 27, continuity between the signal A and the signal B is not ensured as in the example of FIG. 26, and level interpolation is performed so that the gap period is masked to the last. ing.

  FIG. 28 is a diagram illustrating a configuration example of the noise recognition unit 210 according to the embodiment of the present invention. FIG. 28 (a) recognizes noise in the time domain, and FIG. 28 (b) recognizes noise in the frequency domain.

  In the configuration example of FIG. 28A, the noise recognition unit 210 includes a frame generation unit 211, a noise pattern matching unit 212, and a noise pattern holding unit 213.

  The frame generation unit 211 frames the directional signal supplied via the signal line 119 every predetermined time. Here, a frame is a data string composed of a plurality of audio sampling signals. The framed N (N is an integer) audio sampling signals S (n) are supplied to the noise pattern matching unit 212. However, n represents an integer of 1 to N.

  The noise pattern holding unit 213 is a memory that holds the noise pattern W (n). This noise pattern (also referred to as a wavelet) is further read from the noise pattern holding unit 213 as a function W ((n−b) / a) of a and b. Here, a is a scale parameter (where a> 0), and a small value corresponds to noise recognition of a low frequency component. On the other hand, a large scale parameter corresponds to high frequency component noise recognition. Further, b is a shift parameter and represents a shift position (time) at the time of pattern matching with a noise pattern. A wavelet is a function whose signal average value is 0 and is localized around time 0. In the embodiment of the present invention, a function that approximates an actual noise waveform is selected in advance, and a noise pattern holding unit is selected. It shall be held at 213.

The noise pattern matching unit 212 performs convolution while changing the directivity signal S (n) framed by the frame generation unit 211 and the noise pattern W (n) held in the noise pattern holding unit 213 and a and b. By performing the calculation, noise existing in the directional signal is evaluated. The evaluation value Et in this case is calculated by the following formula.

  That is, the evaluation value Et is an index indicating how much the noise pattern W (n) is included in the audio signal S (n), and when there is noise in the directional signal S (n) for each frame. The evaluation value Et becomes large, and the evaluation value Et approaches zero when there is little correlation with noise.

  In the configuration example of FIG. 28B, the noise recognition unit 210 includes a frame generation unit 214, a Fourier transform unit 215, a noise pattern matching unit 216, and a noise pattern holding unit 217.

  Similar to the frame generation unit 211, the frame generation unit 214 converts the directional signal supplied via the signal line 119 into frames every predetermined time. The Fourier transform unit 215 performs Fourier transform on the directivity signal framed by the frame generation unit 214 by FFT (Fast Fourier Transform), and transforms the time signal into the frequency signal F (n).

  The noise pattern holding unit 217 is a memory that holds the noise pattern P (n). The noise pattern P (n) held in the noise pattern holding unit 217 models a frequency distribution when noise is generated.

The noise pattern matching unit 216 obtains the directivity by obtaining the correlation between the directivity signal F (n) converted by the Fourier transform unit 215 and the noise pattern P (n) held in the noise pattern holding unit 213. This is to evaluate noise existing in the sex signal. The evaluation value Ef in this case is calculated by the following equation.

  Here, N is the number of FFT points in one frame. That is, when n is 1 to N and the similarity between the noise pattern and the directivity signal is high, the evaluation value Ef approaches 1, so that if the noise pattern and the directivity signal are equal to or greater than a predetermined threshold value, the patterns are recognized as substantially matching. be able to.

  When noise is recognized in this way, the noise removal period generation unit 220 generates a period determined by the start and end points of the noise generation as the noise removal period. Here, although the method of recognizing noise in each of the time domain and the frequency domain has been described, the recognition rate can be further improved by combining these.

  When a plurality of types of noise are assumed, the above-described noise pattern holding unit 213 and noise pattern holding unit 217 hold noise patterns corresponding to a plurality of types of noise and recognize each noise. It will be.

  In the example of FIG. 20, the description has been made assuming a simple changeover switch as the selection switch 190, but this may be realized by, for example, the following cross-fade switch.

  FIG. 29 is a diagram illustrating a configuration example of a crossfade switch 191 as an example of the selection switch 190 according to the embodiment of the present invention. The cross fade switch 191 includes attenuators 192 and 193, a control coefficient generation unit 194, a coefficient inversion unit 195, and a synthesis unit 196.

  Attenuators 192 and 193 are attenuators that attenuate the input signal in accordance with the control coefficient. The control coefficient of the attenuator 192 is supplied from the control coefficient generation unit 194, and the control coefficient of the attenuator 193 is supplied from the coefficient inversion unit 195.

  The control coefficient generation unit 194 generates a control coefficient for the attenuator 192 based on the noise removal period supplied via the signal line 229. The coefficient inversion unit 195 inverts the output of the control coefficient generation unit 194. That is, the control coefficients of the attenuators 192 and 193 are inverted from each other.

  The synthesizer 196 synthesizes the outputs of the attenuators 192 and 193 and is realized by an adder, for example.

  FIG. 30 is a diagram illustrating a waveform signal example of the crossfade switch 191 in the embodiment of the present invention. When the signal 31 as shown in FIG. 30A is input to the signal line 229, the output signal of the control coefficient generation unit 194 crossfades with a predetermined time constant like the signal 32. On the other hand, the output signal of the coefficient inverting unit 195 is an inverted signal 33 of the signal 32, and similarly crossfades with a predetermined time constant. Therefore, the occurrence of overshoot and ringing can be prevented. Further, the discontinuity of the waveform at the time of switching the outputs of the attenuators 192 and 193 can be absorbed in the sense of hearing, and there is an advantage that it works advantageously for the masking effect.

  FIG. 31 is a diagram illustrating an example of an interpolation signal when the crossfade switch 191 according to the embodiment of the present invention is used. If the level modulation unit 173 outputs an interpolation signal as shown in FIG. 26, when the crossfade switch 191 is used, the signal is crossfade at the transition between the signals A and B and the interpolation signal as shown in FIG. Smooth switching can be realized.

  FIG. 32 is a diagram illustrating a second configuration example of the noise reduction mechanism according to the embodiment of the present invention. Similar to the first configuration example, a directivity signal for noise detection is input to the noise reduction mechanism via the signal line 118, and other directivity signals used as original audio signals are signals. A noise reduction process is performed on the directional signal, which is input via the line 119.

  In this second configuration example, in addition to the first configuration example, a noise removal filter 143, a spectrum envelope generation unit 161, a spectrum coefficient generation unit 162, and a variable filter 163 are further provided. These are assumed to be included in the noise reduction processing unit 370, but are not limited thereto.

  Similar to the noise removal filter 141, the noise removal filter 143 is a filter that removes a noise band from the directional signal from the audio input unit 310. The output of the noise removal filter 143 is supplied to the spectrum envelope generator 161. The noise removal filter 143 can be shared with the noise removal filter 141. In this case, the output of the noise removal filter 141 is supplied to the spectrum envelope generation unit 161.

  The spectrum envelope generator 161 continuously detects the envelope (spectrum envelope) of the frequency spectrum of the directional signal from the voice input unit 310. The spectrum envelope generation unit 161 detects the frequency spectrum by detecting the level for each frequency of the directivity signal (audio signal) by FFT or a plurality of band divisions. The output of the spectrum envelope generator 161 is supplied to the spectrum coefficient generator 162.

  The spectrum coefficient generation unit 162 generates a spectrum coefficient based on the spectrum envelope supplied from the spectrum envelope generation unit 161. In the spectrum coefficient generation unit 162, a spectrum coefficient is generated so as to reproduce the frequency spectrum detected by the spectrum envelope generation unit 161. The output of the spectral coefficient generation unit 162 is supplied to the variable filter 163 via the signal line 168.

  The variable filter 163 performs frequency modulation on the interpolation source signal supplied from the inverse filter 142 in accordance with the spectrum coefficient supplied from the spectrum coefficient generation unit 162. Accordingly, since the interpolation is continuously performed not only for the level modulation in the level modulation unit 173 but also for the frequency component, the gap length can be further increased by the first characteristic.

  Note that, in the second configuration example, the selection switch 190 can be replaced with the cross-fade switch 191 as in the first configuration example.

  FIG. 33 is a diagram illustrating a third configuration example of the noise reduction mechanism according to the embodiment of the present invention. Similar to the first and second configuration examples, a directivity signal for noise detection is input to the noise reduction mechanism via the signal line 118 and the directivity used as the other original audio signal is used. A signal is input via the signal line 119, and noise reduction processing is performed on the directional signal.

  In the third configuration example, a delay unit 120 is provided in addition to the second configuration example, and the outputs subjected to a delay of a predetermined time by the delay unit 120 are the noise removal filters 141 and 143 and the level envelope generation unit 171. Has been supplied to. A signal line 157 from the noise recognition unit 210 is supplied to the variable filter block 140. The variable filter block 140 is a block including a noise removal filter 141, an inverse filter 142, and a noise removal filter 143.

  The noise recognition unit 210 in the third configuration example detects the frequency of the recognized noise and feeds it back to the variable filter block 140. As a method for detecting the noise frequency, the noise frequency can be calculated from the scale parameter a that most closely matches the noise pattern at the time of noise recognition in the time domain of FIG. In addition, when the noise is recognized in the frequency domain of FIG. 28B, the noise frequency can be calculated by detecting the noise peak frequency from the Fourier transform unit 215.

  The noise frequency fed back from the noise recognition unit 210 is used for adjusting the passband or stopband in each filter of the variable filter block 140. Accordingly, for example, by changing the center frequencies fa, fb, and fc in FIG. 23 adaptively according to the noise frequency, it is possible to prevent fluctuations in the noise frequency and continuous noise from a plurality of noise generation sources. It can respond effectively.

  In the third configuration example, since the directivity signal is supplied to the parts other than the noise recognition unit 210 via the delay unit 120, the pass band or the stop band can be adjusted in real time according to the result of the noise recognition. .

  Note that the third configuration example is the same as the first and second configuration examples in that the selection switch 190 can be replaced with the cross-fade switch 191.

  Next, the operation of the imaging apparatus according to the embodiment of the present invention will be described with reference to the drawings.

  FIG. 34 is a diagram showing an example of a basic processing procedure of the noise reduction method in the sound collection device 300 according to the embodiment of the present invention. This processing procedure example is common to the first to third configuration examples described above.

  First, noise recognition processing is performed in the noise recognition unit 210 (step S910). As a result, the noise removal period generator 220 generates a noise removal period. When the noise removal period is satisfied (step S920), the directivity signal supplied from the noise removal filter 141 via the signal line 149 is selected by the selection switch 190 (step S930). On the other hand, when it does not correspond to the noise removal period (step S920), the directivity signal supplied from the audio input unit 310 via the signal line 119 is selected (step S940). The above process is repeated.

  As described above, according to the embodiment of the present invention, the noise detection unit 360 performs noise detection based on the directivity signal generated by the directivity generation unit 330, and the noise reduction processing unit is performed on the directivity signal according to the result. Noise removal can be performed at 370. The directivity signal used for noise in the noise detection unit 360 corresponds to a noise vector, and noise can be detected efficiently. In addition, the noise reduction processing unit 370 specifies a noise removal period from the noise recognized by the noise recognition unit 210, selects a signal from which noise has been removed by the noise removal filter 141 during the noise removal period, and performs other periods. By controlling the selection switch 190 so as to select a directional signal from which noise is not removed, it is possible to realize noise reduction processing in consideration of human hearing. Further, according to the embodiment of the present invention, it is possible to reduce noise that continues for a long time by synthesizing the interpolation signal in the noise removal period.

  In the embodiment of the present invention, the example in which the recording stream signal is generated by scanning the directional signal and the sampling frequency is changed by the upsampling process and the downsampling process has been described. Each directivity signal may be handled individually without performing the above.

  In the embodiment of the present invention, an example in which a 5.1 channel surround signal is assumed has been described. However, the present invention is not limited to this, and the present invention is not limited to this case even when the number of channels is appropriately increased or decreased. The present invention can be similarly implemented without departing from the object of the invention.

  Further, the embodiment of the present invention shows an example for embodying the present invention, and has a corresponding relationship with the invention specific matter in the claims as shown below, but is not limited thereto. However, various modifications can be made without departing from the scope of the present invention.

  That is, in claim 1, the voice input unit corresponds to the voice input unit 310, for example. The directivity generation means corresponds to the directivity generation unit 330, for example. The noise removing unit corresponds to the noise removing filter 141, for example. Further, the noise recognition means corresponds to the noise recognition unit 210, for example. Further, the noise removal period generation unit corresponds to the noise removal period generation unit 220, for example. The selection means corresponds to the selection switch 190, for example.

  Further, in claim 2, the bidirectional microphone corresponds to, for example, the bidirectional microphones 412 and 413. An omnidirectional microphone corresponds to the omnidirectional microphone 411, for example.

  Further, in claim 3, the omnidirectional microphone corresponds to, for example, the omnidirectional microphones 431 to 434.

  Further, in claim 4, unidirectional microphones correspond to, for example, unidirectional microphones 452 and 453. The bidirectional microphone corresponds to, for example, the bidirectional microphone 451.

  Further, in claim 5, the rotation coefficient generation means corresponds to, for example, the rotation coefficient generation unit 350.

  In claim 11, the voice input means corresponds to the voice input unit 310, for example. The directivity generation means corresponds to the directivity generation unit 330, for example. The noise removing unit corresponds to the noise removing filter 141, for example. The signal interpolation means includes, for example, an interpolation source signal generation unit 130, an inverse filter 142, a noise removal filter 143, a spectrum envelope generation unit 161, a spectrum coefficient generation unit 162, a variable filter 163, a level envelope generation unit 171, and a level coefficient generation unit 172. This corresponds to at least a partial combination of the level modulation unit 173 and the synthesis unit 180. Further, the noise recognition means corresponds to the noise recognition unit 210, for example. Further, the noise removal period generation unit corresponds to the noise removal period generation unit 220, for example. The selection means corresponds to the selection switch 190, for example.

  Further, in claim 12, the interpolation source signal generation means corresponds to the interpolation source signal generation unit 130, for example. Further, the out-of-interpolation removing unit corresponds to the inverse filter 142, for example. The level envelope generation means corresponds to the level envelope generation unit 171, for example. The level coefficient generation means corresponds to the level coefficient generation unit 172, for example. The level modulation means corresponds to the level modulation unit 173, for example. A synthesizing unit corresponds to the synthesizing unit 180, for example.

  Further, in claim 15, the interpolation source signal generation means corresponds to the interpolation source signal generation unit 130, for example. Further, the out-of-interpolation removing unit corresponds to the inverse filter 142, for example. The spectrum envelope generation means corresponds to, for example, the spectrum envelope generation unit 161. The spectral coefficient generation unit corresponds to, for example, the spectral coefficient generation unit 162. The spectrum modulation means corresponds to the variable filter 163, for example. The level envelope generation means corresponds to the level envelope generation unit 171, for example. The level coefficient generation means corresponds to the level coefficient generation unit 172, for example. The level modulation means corresponds to the level modulation unit 173, for example. A synthesizing unit corresponds to the synthesizing unit 180, for example.

  The processing procedure described in the embodiment of the present invention may be regarded as a method having a series of these procedures, and a program for causing a computer to execute these series of procedures or a recording medium storing the program May be taken as

It is a figure which shows one structural example of the sound collection apparatus 300 in embodiment of this invention. It is a figure which shows arrangement | positioning and directivity of 5.1 surround sound source. It is a figure which shows an example of the vector amount extraction in embodiment of this invention. It is a figure which shows the example of the polar pattern by the sound collection device in embodiment of this invention. It is a figure which shows the 1st example of arrangement | positioning of the microphone in embodiment of this invention. It is a figure which shows the synthesis example of the sound source of the microphone by the 1st example of arrangement | positioning by embodiment of this invention. It is a figure which shows the rotation coefficient in embodiment of this invention. It is a figure which shows an example of the directivity characteristic of the microphone in the 1st example of arrangement | positioning by embodiment of this invention. It is a figure which shows the 2nd example of arrangement | positioning of the microphone in embodiment of this invention. It is a figure which shows the production | generation example of the directional characteristic of the microphone in the 2nd example of arrangement | positioning by embodiment of this invention. It is a figure which shows the synthesis example of the sound source of the microphone in the 2nd example of arrangement | positioning by embodiment of this invention. It is a figure which shows the 3rd example of arrangement | positioning of the microphone in embodiment of this invention. It is a figure which shows the synthesis example of the sound source of the microphone in the 3rd example of arrangement | positioning by embodiment of this invention. It is a figure which shows the production | generation example of the directional characteristic of the microphone in the 3rd example of arrangement | positioning by embodiment of this invention. It is a figure which shows the example of the rotation angle of directivity in embodiment of this invention. It is a figure which shows the example of the content of the directional stream signal in embodiment of this invention. It is a figure which shows the relationship between the directional stream signal and sampling period in embodiment of this invention. It is a figure which shows the example of 1 structure of the directivity production | generation part 330 in embodiment of this invention. It is a figure which shows one structural example of the downsampling mechanism in embodiment of this invention. It is a figure which shows the 1st structural example of the noise reduction mechanism in embodiment of this invention. It is a figure for demonstrating the masking phenomenon utilized in embodiment of this invention. It is a figure which shows one structural example of the interpolation source signal generation part 130 in embodiment of this invention. It is a figure which shows the frequency characteristic example of the noise removal filter 141 and the inverse filter 142 in embodiment of this invention. It is a figure which shows the example of 1 structure of the level envelope production | generation part 171 in embodiment of this invention. It is a figure which shows an example of the process in the level envelope production | generation part 171 in embodiment of this invention. It is a figure which shows an example of the interpolation signal in embodiment of this invention. It is a figure which shows the other example of the interpolation signal in embodiment of this invention. It is a figure which shows the example of 1 structure of the noise recognition part 210 in embodiment of this invention. It is a figure which shows the structural example of the crossfade switch 191 as an example of the selection switch 190 in embodiment of this invention. It is a figure which shows the waveform signal example of the cross fade switch 191 in embodiment of this invention. It is a figure which shows the example of the interpolation signal at the time of using the cross fade switch 191 in embodiment of this invention. It is a figure which shows the 2nd structural example of the noise reduction mechanism in embodiment of this invention. It is a figure which shows the 3rd structural example of the noise reduction mechanism in embodiment of this invention. It is a figure which shows the example of a basic process sequence of the noise reduction method in the sound collection device 300 by embodiment of this invention.

Explanation of symbols

DESCRIPTION OF SYMBOLS 300 Sound collecting device 310 Audio | voice input part 320 Amplifier 330 Directivity production | generation part 331 Upsampling part 332 Interpolation filter 333 Directivity production | generation part 340 Timing production | generation part 350 Rotation coefficient production | generation part 360 Noise detection part 370 Noise reduction process part 371 Directivity Direction extraction unit 372 Decimation filter 373 Downsampling unit 380 Coding processing unit 390 Recording / playback unit 411 Nondirectional microphone 412 413 Bidirectional microphone 422, 423 Level variable unit 426 Addition synthesis unit 431-434 Nondirectional microphone 441 Addition Unit 442, 443 subtraction unit 444, 445 level variable unit 446 addition synthesis unit 451 bi-directional microphone 452, 453 unidirectional microphone 461-463 level variable unit 66 adding and combining unit 506, 507 bi-directional signal

Claims (16)

Voice input means for inputting a plurality of surrounding voice signals;
Directivity generation means for generating a first directivity signal having directivity in a first direction and a second directivity signal having directivity in a second direction based on the plurality of audio signals;
Noise removing means for removing a noise band from the first directional signal;
Noise recognition means for recognizing noise included in the second directional signal;
Noise removal period generation means for generating a signal indicating a noise removal period according to the recognized noise generation period;
When the noise removal period is indicated, the output of the noise removal means is selected, and when the noise removal period is not indicated, the first directivity signal is selected. And a sound collecting device.   2. The sound collecting apparatus according to claim 1, wherein the voice input unit includes a plurality of bidirectional microphones and one omnidirectional microphone.   The sound collection device according to claim 1, wherein the voice input unit includes a plurality of omnidirectional microphones.   2. The sound collection device according to claim 1, wherein the voice input unit includes a plurality of unidirectional microphones and one bi-directional microphone. A rotation coefficient generating means for generating a rotation coefficient indicating a predetermined direction;
The directivity generation means generates the first directivity signal if the direction indicated by the rotation coefficient is the first direction, and generates the first directivity signal if the direction indicated by the rotation coefficient is the second direction. The sound collecting device according to claim 1, wherein two directional signals are generated.   The noise recognition means performs the noise recognition using an output obtained by convolution of a wavelet signal having a mean value of zero in a predetermined period approximated to the noise as a waveform and the second directivity signal as an evaluation value. The sound collecting device according to claim 1.   The noise recognition means performs the noise recognition using an evaluation value as a correlation between a pattern signal approximated to a frequency spectrum of the noise and the second directional signal subjected to Fourier transform. Sound collecting device.   2. The sound collecting device according to claim 1, wherein the noise removing unit is a filter for removing a noise band.   9. The sound collecting device according to claim 8, wherein the noise removing unit adaptively changes a removal band and a pass band of the filter based on a frequency of noise recognized by the noise recognizing unit.   2. The sound collection device according to claim 1, wherein the selection means is a cross-fade switch. Voice input means for inputting a plurality of surrounding voice signals;
Directivity generation means for generating a first directivity signal having directivity in a first direction and a second directivity signal having directivity in a second direction based on the plurality of audio signals;
Noise removing means for removing a noise band from the first directional signal;
Signal interpolation means for performing interpolation on the signal from which the noise band has been removed;
Noise recognition means for recognizing noise included in the second directional signal;
Noise removal period generation means for generating a signal indicating a noise removal period according to the recognized noise generation period;
When the noise removal period is indicated, the output of the signal interpolation means is selected, and when the noise removal period is not indicated, the first directivity signal is selected. And a sound collecting device. The signal interpolation means includes
Interpolation source signal generation means for generating an interpolation source signal for the interpolation;
Non-interpolation removal means for removing the noise source band other than the noise band from the interpolation source signal;
Level envelope generating means for generating a level envelope of the first directional signal;
Level coefficient generating means for generating a level coefficient for the interpolation based on the level envelope;
Level modulation means for modulating the output of the non-interpolation removal means based on the level coefficient;
12. The sound collecting apparatus according to claim 11, further comprising a combining unit that combines the output of the noise removing unit and the output of the level modulating unit and outputs the combined result to the selecting unit.   13. The sound collecting device according to claim 12, wherein the level modulation means further modulates the output of the extrapolation removal means based on a level masked on human hearing.   The interpolation source signal generation means includes a plurality or a single periodic signal having a predetermined waveform and a predetermined period, a white noise signal having a uniform level in a voice band, or a predetermined mixture of the periodic signal and the white noise signal. 13. The sound collecting device according to claim 12, wherein one of the mixed signals based on the ratio is generated. The signal interpolation means includes
Interpolation source signal generation means for generating an interpolation source signal for the interpolation;
Non-interpolation removal means for removing the noise source band other than the noise band from the interpolation source signal;
A spectrum envelope generating means for generating a frequency spectrum envelope of the output of the noise removing means;
Spectral coefficient generating means for generating a spectral coefficient for the interpolation based on the spectral envelope;
Spectrum modulating means for modulating the output of the out-of-interpolation removing means based on the spectral coefficient;
Level envelope generating means for generating a level envelope of the first directional signal;
Level coefficient generating means for generating a level coefficient for the interpolation based on the level envelope;
Level modulation means for modulating the output of the spectrum modulation means based on the level coefficient;
12. The sound collecting apparatus according to claim 11, further comprising a combining unit that combines the output of the noise removing unit and the output of the level modulating unit and outputs the combined result to the selecting unit.   16. The filter according to claim 15, wherein the noise removal unit and the non-interpolation removal unit are filters that adaptively change a removal band and a pass band based on a frequency of noise recognized by the noise recognition unit. Sound collection device.

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