Patent Publication Number: US-8995681-B2

Title: Audio processing apparatus with noise reduction and method of controlling the audio processing apparatus

Description:
BACKGROUND OF THE INVENTION 
     1. Field of the Invention 
     The present invention relates to an audio processing apparatus and a method of controlling the audio processing apparatus. 
     2. Description of the Related Art 
     Video cameras, IC recorders, and the like are conventionally known as audio processing apparatuses. In these audio processing apparatuses, an audio signal acquired from a microphone may contain noise due to the influence of wind. As a countermeasure, some apparatuses provide a gain controller before an A/D converter to prevent an audio signal that has passed through the A/D converter from being saturated, and also remove low-frequency components to reduce wind noise in the audio signal that has passed through the A/D converter. For example, Japanese Patent Laid-Open No. 2008-129107 discloses a method of obtaining a high-quality audio by providing a gain controller before an A/D converter and also providing a gain controller after a low-frequency removing unit for wind noise processing. 
     However, in the conventional technique disclosed in Japanese Patent Laid-Open No. 2008-129107, the quantization error may become large upon gain control after wind noise processing. For example, according to the method of Japanese Patent Laid-Open No. 2008-129107, when the gain controller increases the gain, the quantization error of the above-described A/D converter becomes large. 
     SUMMARY OF THE INVENTION 
     The present invention provides a high-quality audio by suppressing an increase in the quantization error by gain control after wind noise processing. 
     According to an aspect of the present invention, an audio processing apparatus includes a first audio pickup unit, a second audio pickup unit including an audio resistor provided to cover a sound receiving portion to suppress external wind introduction while passing an external audio, a first A/D converter that digitizes an output signal from the first audio pickup unit, a second A/D converter that digitizes an output signal from the second audio pickup unit, a level controller that controls at least one of a signal level of an output signal of the first A/D converter and a signal level of an output signal of the second A/D converter, a first filter that attenuates a signal having a frequency lower than a first cutoff frequency of the output signal of the first A/D converter, a third filter that attenuates a signal having a frequency higher than a second cutoff frequency of the output signal of the second A/D converter, an adder that adds an output signal of the first filter and an output signal of the third filter to output an audio with reduced wind noise, and a second filter provided between the first audio pickup unit and the first A/D converter to attenuate a signal having a frequency lower than a third cutoff frequency for suppressing the wind noise. 
     According to the present invention, it is possible to provide a high-quality audio by suppressing an increase in the quantization error by gain control after wind noise processing. 
     Further features and aspects of the present invention will become apparent from the following detailed description of exemplary embodiments with reference to the attached drawings. 
    
    
     
       BRIEF DESCRIPTION OF THE DRAWINGS 
       The accompanying drawings, which are incorporated in and constitute a part of the specification, illustrate exemplary embodiments, features, and aspects of the invention and, together with the description, serve to explain the principles of the invention. 
         FIG. 1  is a block diagram showing the arrangement of an audio recorder according to an embodiment; 
         FIGS. 2A and 2B  are perspective and sectional views, respectively, showing an image capture device; 
         FIGS. 3A to 3F  are graphs showing examples of the frequency characteristic of a microphone; 
         FIGS. 4A to 4D  are views for explaining the attachment structure of microphones; 
         FIG. 5  is a block diagram showing the arrangement of a reverberation suppressor; 
         FIGS. 6A to 6D  are timing charts showing the operation of a wind-detector according to wind noise; 
         FIGS. 7A to 7D  are views showing the arrangements and operations of a mixer; 
         FIGS. 8A to 8D  are graphs showing the operation sequences of a switch, variable filters, ad a variable gain; 
         FIG. 9  is a timing chart for explaining wind noise processing when no HPF exists; 
         FIG. 10  is a timing chart for explaining wind noise processing when an HPF exists; 
         FIGS. 11A and 11B  are block diagrams showing other examples of the audio processing apparatus; 
         FIG. 12  is a perspective view showing an image capture device according to the second embodiment; and 
         FIG. 13  is a block diagram showing the arrangement of an audio processing apparatus according to the second embodiment. 
     
    
    
     DESCRIPTION OF THE EMBODIMENTS 
     Various exemplary embodiments, features, and aspects of the invention will be described in detail below with reference to the drawings. 
     First Embodiment 
     An audio recorder serving as an audio processing apparatus and an image capture device including the audio recorder according to the first embodiment of the present invention will be described below with reference to  FIGS. 1 to 11A  and  11 B. 
       FIG. 1  is a block diagram showing the arrangement of the audio recorder according to this embodiment.  FIGS. 2A and 2B  are perspective and sectional views, respectively, showing the image capture device (camera) including the audio recorder shown in  FIG. 1 . Reference numeral  1  denotes an image capture device;  2 , a lens attached to the image capture device  1 ;  3 , a body of the image capture device  1 ;  4 , an optical axis of the lens;  5 , a photographing optical system; and  6 , an image sensor. Reference numeral  30  denotes a release button; and  31 , an operation button. A first microphone  7   a  and a second microphone  7   b  are provided in the image capture device  1 . Opening portions  32   a  and  32   b  are provided in the body  3  for the microphones  7   a  and  7   b , respectively. An audio resistor  41  for suppressing wind introduction while passing external audio is pasted to the opening portion  32   b  to cover the sound receiving portion of the microphone  7   b . The audio resistor  41  can also be formed by making the body  3  have an uneven thickness or using an extra part, as will be described later. The image capture device  1  can simultaneously perform image acquisition and audio recording using the microphones  7   a  and  7   b.    
     The moving image shooting operation of the image capture device  1  will be explained. When the user presses a live view button (not shown) before moving image shooting the image on the image sensor  6  is displayed on a display device provided in the image capture device  1  in real time. In synchronism with the operation of a moving image shooting button, the image capture device  1  obtains object information from the image sensor  6  at a set frame rate and audio information from the microphones  7   a  and  7   b  simultaneously, and synchronously records these pieces of information in a memory (not shown). Shooting ends in synchronism with the operation of the moving image shooting button. 
     The arrangement of an audio processing apparatus  51  will be described with reference to  FIG. 1 . Reference numeral  52  denotes an analog high-pass filter (HPF) configured to change the cutoff frequency;  53 , a reverberation suppressor formed from, for example, a reverberation suppression adaptive filter;  54   a  and  54   b , first A/D converters (ADCs) that digitize the signals output from the microphones;  55 , a first delay device (DL)  55 ; and  56   a  and  56   b , DC component cutting HPFs. 
     Reference numeral  61  denotes an automatic level controller (ALC). The ALC  61  includes variable gains  62   a  and  62   b  for level control, and a level controller  63 . 
     A mixer  71  mixes the signal of the first microphone  7   a  and signal of the second microphone  7   b . The mixer  71  includes a low-pass filter (LPF)  72 , an HPF  73  configured to change the cutoff frequency, a gain multiplier  74 , and an adder  75 . 
     Reference numeral  81  denotes a wind-detector. The wind-detector  81  includes bandpass filters (BPFs)  82   a  and  82   b , a subtracter  83 , a second A/D converter (ADC)  84 , a second delay device  85 , and a level detector  86 . 
     Reference numeral  87  denotes a switch that controls the reverberation suppressor  53 ;  88 , a switch that controls the mixer  71 ; and  89 , a mode switching operation unit. 
     Needless to say, a high-pass filter attenuates a signal having a frequency lower than a predetermined frequency but does not attenuate a signal having a frequency higher than the predetermined frequency. Thus, the high-pass filter attenuates, out of an input signal, signal components having frequencies lower than a predetermined frequency more than those having frequencies higher than the predetermined frequency. The predetermined frequency is called a cutoff frequency. Similarly, a low-pass filter attenuates a signal having a frequency higher than a predetermined frequency but does not attenuate a signal having a frequency lower than the predetermined frequency. Thus, the low-pass filter attenuates, out of an input signal, signal components having frequencies higher than a predetermined frequency more than those having frequencies lower than the predetermined frequency. The predetermined frequency is called a cutoff frequency. A bandpass filter attenuates signals outside a predetermined frequency range but does not attenuate signals within the predetermined frequency range. Thus, the bandpass filter attenuates signals outside a predetermined frequency range more than those within the predetermined frequency range. In other words, these filters extract signals having desired frequencies. 
     Referring to  FIGS. 1 ,  2 A, and  2 B, the opening portions  32   a  and  32   b  for the microphones are provided in the body  3 . The audio resistor  41  that covers the second microphone  7   b  is provided on the opening portion  32   b  to mask movement of air from the outside of the apparatus to the second microphone  7   b . On the other hand, the opening portion  32   a  is not provided with such an audio resistor so that the first microphone  7   a  can faithfully acquire an object sound. The audio resistor  41  is provided in tight contact with the body  3 . The movement of air is here assumed to be air movement by wind. For example, a material such as porous PTFE that allows air to move more slowly than air moved by wind but does not allow the wind to pass through can also be used as the audio resistor. 
     In the audio processing apparatus  51 , the signal from the first microphone  7   a  is processed by the HPF  52  and then undergoes analog/digital conversion (A/D conversion) of the ADC  54   a . The first delay device  55  delays the output from the ADC  54   a  by an appropriate amount. On the other hand, in the audio processing apparatus  51 , the signal from the second microphone  7   b  is A/D-converted by the ADC  54   b  and then undergoes reverberation suppression of the reverberation suppressor  53 . The operation of the reverberation suppressor  53  and how to cause the first delay device  55  to apply a delay will be described later. 
     The outputs from the first delay device  55  and the ADC  54   b  are processed by the DC component cutting HPFs  56   a  and  56   b , respectively. The HPFs  56   a  and  56   b  aim at removing the offset of the analog part and need only remove components below the audible range from the DC. To do this, the cutoff frequency of the HPFs  56   a  and  56   b  is set to, for example, about 10 Hz. 
     The outputs from the HPFs  56   a  and  56   b  are input to the ALC  61  and undergo gain control of the variable gains  62   a  and  62   b . At this time, the gain of at least one of the variable gains  62   a  and  62   b  is controlled such that, for example, the two signal levels of, 2 kHz that is a frequency lower than that of the HPF  56  become identical. The level controller  63  receives the outputs from the variable gains  62   a  and  62   b  and appropriately controls the levels so as to effectively use the dynamic range without causing saturation. At this time, the level controller  63  performs level control not to cause saturation of a larger one of the outputs from the variable gains  62   a  and  62   b.    
     The outputs from the variable gains  62   a  and  62   b  are input to the mixer  71 . The output from the variable gain  62   a  is passed through the HPF  73  and sent to the adder  75 . On the other hand, the output from the variable gain  62   b  is sent to the adder  75  via the LPF  72  and the variable gain  74 . The output mixed by the adder  75  is output as the audio after wind noise processing. 
     The output from the first microphone  7   a  and the output from the reverberation suppressor  53  are input to the BPFs  82   a  and  82   b  of the wind-detector  81 , respectively. The BPFs  82   a  and  82   b  aim at passing components within the range where the object sound can faithfully be acquired by the second microphone  7   b . Thus, the passband is set to, for example, about 30 Hz to 1 kHz. However, the upper limit set value of the frequency can be changed by the structure of the audio resistor  41  or the like. Details will be described later together with the frequency characteristic of the second microphone  7   b.    
     The output from the BPF  82   a  is A/D-converted by the second ADC  84  and sent to the second delay device  85 . How to cause the second delay device  85  to apply a delay will be described later together with the operation of the reverberation suppressor  53 . 
     The subtracter  83  calculates the difference between the outputs from the second delay device  85  and the output from the BPF  82   b  and sends the result to the level detector  86 . The operation of the level detector  86  will be described later. The level detector  86  determines the strength of wind, and the switch  87  is controlled to switch feedback to the reverberation suppressor  53 . The detection result of the level detector  86  is also used to control the switch  88  for controlling the mixer  71 . When the user sets the mode switching operation unit  89  to OFF, the switch  88  operates to always select processing in the windless state to be described later. On the other hand, when the user sets the mode switching operation unit  89  to Auto, the switch  88  operates to change the cutoff frequencies of the HPF  52  and the HPF  73  and the variable gain  74  in accordance with the wind strength determined by the level detector  86 . Details of this processing will be described later. 
     The effects and desired characteristics of the audio resistor  41  and wind noise reduction will be explained with reference to  FIGS. 1 ,  3 A to  3 F, and  4 A to  4 D.  FIGS. 3A to 3F  are graphs schematically showing the frequency characteristic of the microphone. The abscissa represents the frequency, and the ordinate represents the gain.  FIG. 3A  shows the object sound acquisition characteristic of the first microphone  7   a .  FIG. 3B  shows the object sound acquisition characteristic of the second microphone  7   b .  FIG. 3C  shows the wind noise acquisition characteristic of the first microphone  7   a .  FIG. 3D  shows the wind noise acquisition characteristic of the second microphone  7   b .  FIG. 3E  shows the object sound acquisition characteristic of the output of the mixer  71 .  FIG. 3F  shows the wind noise acquisition characteristic of the output of the mixer  71 . To clarify the characteristic difference between the first microphone  7   a  and the second microphone  7   b , the characteristics of the first microphone  7   a  are indicated by the broken lines in FIGS.  3 B and  3 D. In  FIGS. 3A and 3B , f 0  represents the structural cutoff frequency by the audio resistor  41 , and f 1  represents the cutoff frequency of the LPF  72  and the HPF  73  in the mixer  71  shown in  FIG. 1 . 
     As shown in  FIG. 3A , the object sound acquisition characteristic of the first microphone  7   a  is preferably flat in the audible range. This allows to faithfully acquire the object sound. As shown in  FIG. 3B , the second microphone  7   b  has a different characteristic because the audio resistor  41  is provided to mask movement of air from the object. The second microphone  7   b  relatively faithfully passes the audio signal at a frequency lower than the cutoff frequency by the audio resistor  41 . This is because the sound that is a compressional wave of air excites the audio resistor  41 , and the audio resistor  41  thus excites the air in the apparatus in the same way. On the other hand, the second microphone  7   b  masks the audio signal at a frequency higher than the cutoff frequency by the audio resistor  41 . This is because although the sound that is a compressional wave of air excites the audio resistor  41 , the density is inverted before the audio resistor  41  starts vibrating, and the air cannot move. Thus, the audio resistor  41  suppresses wind noise and acts as a structural low-pass filter for an audio other than the wind noise. The frequency f 0  at which the structural cutoff begins will be referred to as the cutoff frequency of the audio resistor  41 . 
     The power of wind noise is known to concentrate to the lower frequency range. For example, as for the power of wind noise in the first microphone  7   a , a characteristic that rises from about 1 kHz to the lower frequency side is obtained in many cases, as shown in  FIG. 3C . Even if the shape is different from that shown in  FIG. 3C , low-frequency components (equal to or lower than 500 Hz) are dominant in the wind noise. As shown in  FIG. 3D , the rise of the low-frequency components of wind noise is small in the second microphone  7   b . Near the first microphone  7   a , a large atmospheric pressure difference is readily generated because of a turbulent flow or the like. For the second microphone  7   b , however, such a large atmospheric pressure difference is not caused by a turbulent flow or the like because the audio resistor  41  is provided to mask movement of air from the object. This is the reason why the rise of the low-frequency components of wind noise is small in the output of the second microphone  7   b.    
     Consider processing of these signals by the mixer  71 . As described above with reference to  FIG. 1 , the signal of the first microphone  7   a  is processed by the HPF  73 . This corresponds to cutting a portion  91  in  FIG. 3A  and a portion  93  in  FIG. 3C . The signal of the second microphone  7   b  is processed by the LPF  72 . This corresponds to cutting a portion  92  in  FIG. 3B  and a portion  94  in  FIG. 3D . When passing through the adder  75 , an object sound characteristic as shown in  FIG. 3E  is obtained, and a wind noise characteristic as shown in  FIG. 3F  is obtained. The portions  91 ,  92 ,  93 , and  94  are dominant at portions  91   a ,  92   a ,  93   a , and  94   a  shown in  FIGS. 3E and 3F . Note that the expression “dominant” is used because the counterpart is not necessarily zero because of the characteristics of the LPF  72  and the HPF  73 . As is apparent from  FIGS. 3E and 3F , the output of the mixer  71  has a flat object sound characteristic in the audible range and a wind noise characteristic equal to the characteristic of the microphone provided with the audio resistor  41 . 
       FIGS. 4A to 4D  illustrate examples of the attachment structure of the microphones. Referring to  FIGS. 4A to 4D , reference numerals  33   a  and  33   b  denote holding elastic bodies of the first microphone  7   a  and the second microphone  7   b  respectively; and  34 , a sleeve that holds the second microphone  7   b  and the audio resistor  41 . 
       FIG. 4A  shows an example in which the audio resistor  41  is pasted outside the body  3 . In the example of  FIG. 4A , the audio resistor  41  can be pasted after the apparatus has been assembled. This enables to improve the assembling efficiency. 
       FIG. 4B  shows an example in which the audio resistor  41  is pasted inside the body  3 . In the example of  FIG. 4B , since the audio resistor  41  is not exposed to the outside the body  3 , a fine outer appearance can be obtained. 
       FIG. 4C  shows an example in which part of the body  3  also functions as the audio resistor  41 . In the example of  FIG. 4C , the part of the body  3  serving as the audio resistor  41  is made so thin as to be vibrated by a sound wave. In the example of  FIG. 4C , since it is unnecessary to paste the audio resistor  41  to the body  3 , and the number of parts can be reduced, a fine outer appearance can be obtained. In the example of  FIG. 4C , however, since the body  3  and the audio resistor  41  are integrated, the degree of freedom of design generally decreases (the strength of the body  3  may be limited by the thickness of the portion that forms the audio resistor  41 , resulting in difficulty in meeting the requirements simultaneously). 
       FIG. 4D  shows an example in which the sufficiently rigid sleeve  34  holds the second microphone  7   b  and the audio resistor  41 . The sleeve  34  preferably has a primary resonance frequency sufficiently higher than the band of the frequency to be acquired by the second microphone  7   b  (this means that the resonance frequency of the sleeve  34  is higher than f 0  in  FIGS. 3A and 3B ). In the example of  FIG. 4D , the audio resistor  41  is attached to the highly rigid sleeve  34 . It is therefore possible to obtain a desired audio signal in the passband (at a frequency lower than f 0  in  FIGS. 3A and 3B ) without being affected by the unnecessary resonance of the attachment structure. 
     The reverberation suppressor  53  will be described next with reference to  FIGS. 1 and 5 . Since the second microphone  7   b  is covered by the audio resistor  41 , reverberation may occur in the closed space. In this embodiment, the reverberation suppressor  53  is provided to suppress such reverberation. 
       FIG. 5  shows the detailed arrangement of the reverberation suppressor  53 . The reverberation suppressor  53  is formed from an adaptive filter. This adaptive filter estimates and learns the filter coefficient so as to minimize the output of the subtracter  83 , Thus, the difference between the output signal of the first microphone  7   a  and the output signal of the second microphone  7   b , which represents the level of wind noise, as will be described below in detail. The reverberation component generated in the closed space between the audio resistor  41  and the second microphone  7   b  and contained in the output signal of the second microphone  7   b  is thus suppressed. Using such an adaptive filter makes it possible to appropriately perform processing even if the reverberation generation state changes due to the change of the user&#39;s camera grip state or the change in the temperature. 
     The principle of reverberation suppression will briefly be described. Let s be the object sound, g 1  be the object sound acquisition characteristic of the first microphone  7   a , g 2  be the object sound acquisition characteristic of the second microphone  7   b , and r be the influence of reverberation. The object sound acquisition characteristics g 1  and g 2  equal the inverse Fourier transformation results of the characteristics in the frequency space shown in  FIGS. 3A to 3F . A signal x 1  of the first microphone  7   a  and a signal x 2  of the second microphone  7   b  obtained under an environment with reverberation in the second microphone  7   b  are given by
 
 x 1= s*g 1
 
 x 2= s*g 2* r   (1)
 
where * is an operator representing convolution. As described with reference to  FIGS. 3A to 3F , the first microphone  7   a  and the second microphone  7   b  can acquire similar object sounds at a frequency lower than f 0 . As shown in  FIG. 1 , the BPFs  82   a  and  82   b  extract only components in an appropriate band. Thus, the BPFs pass frequencies lower than f 0  in  FIGS. 3A to 3F  within the audible range. The human auditory sense exhibits an extremely low sensitivity to a band of 50 Hz or less because of its characteristic. For further details, see A characteristic curve or the like. Hence, the BPFs  82   a  and  82   b  are designed to pass frequencies of, for example, 30 Hz to 1 kHz. Letting BPF be the BPFs  82   a  and  82   b , and x 1 _BPF and x 2 _BPF be the signals that have passed through the BPFs,
 
 x 1_BPF= s*g 1*BPF
 
 x 2_BPF= s*g 2* r *BPF
 
 g 1*BPF= g 2*BPF  (2)
 
holds. Holding g 1 ≠g 2 , and g 1 *BPF≠g 2 *BPF is equivalent to allowing the first microphone  7   a  and the second microphone  7   b  to acquire similar object sounds at a frequency lower than f 0 . As is apparent from equations (2), identical signals are input to the subtracter  83  in  FIG. 1  when the influence r of reverberation is absent. The influence of reverberation can be reduced by operating the adaptive filter using x 1 _BPF=d as the desired response and x 2 _BPF=u as the input, as can be seen from equations (2).
 
     When the filter of the reverberation suppressor  53  is expressed as h, an adaptive filter output y is given by 
                     y   ⁡     (   n   )       =       h   *   u     =         ∑     i   =   0     M     ⁢         h   n     ⁡     (   i   )       ⁢     u   ⁡     (     n   -   i     )           =       ∑     i   =   0     M     ⁢         h   n     ⁡     (   i   )       ⁢   x2_BPF   ⁢     (     n   -   i     )                     (   3   )               
where n indicates the signal of the nth sample, M is the filter order of the reverberation suppressor  53 , and the subscript of h indicates the value of a filter h of the nth sample. As the input u, x 2 _BPF is used.
 
     In addition, x 1 _BPF=d is used as the desired response. Hence, an error signal e is expressed as 
     
       
         
           
             
               
                 
                   
                     e 
                     ⁡ 
                     
                       ( 
                       n 
                       ) 
                     
                   
                   = 
                   
                     
                       
                         d 
                         ⁡ 
                         
                           ( 
                           n 
                           ) 
                         
                       
                       - 
                       
                         y 
                         ⁡ 
                         
                           ( 
                           n 
                           ) 
                         
                       
                     
                     = 
                     
                       
                         x1_BPF 
                         ⁢ 
                         
                           ( 
                           n 
                           ) 
                         
                       
                       - 
                       
                         
                           ∑ 
                           
                             i 
                             = 
                             0 
                           
                           M 
                         
                         ⁢ 
                         
                           
                             
                               h 
                               n 
                             
                             ⁡ 
                             
                               ( 
                               i 
                               ) 
                             
                           
                           ⁢ 
                           x2_BPF 
                           ⁢ 
                           
                             ( 
                             
                               n 
                               - 
                               i 
                             
                             ) 
                           
                         
                       
                     
                   
                 
               
               
                 
                   ( 
                   4 
                   ) 
                 
               
             
           
         
       
     
     Various adaptive algorithms have been proposed. For example, the update equation of h by the LMS algorithm is given by
 
 h   n+1 ( i )= h   n ( i )+μ e ( n ) u ( n−i ) ( i= 0, 1 , . . . M )  (5)
 
where μ is the step size parameter. According to the above-described method, an appropriate initial value h is given and updated using equation (5), thereby making u closer to d. Thus, the influence r is reduced, and x 1 _BPF=x 2 _BPF almost holds. At this time, |h*r|=1 holds in the passband of the BPF. However, in an environment where the wind noise is dominant, updating of equation (5) is not correctly performed. Hence, the estimation learning of the adaptive filter is stopped by the switch  87 . The control sequence of the switch  87  will be described later together with the operation of the wind-detector  81 .
 
     As described above, the reverberation suppressor  53  suppresses reverberation. In the reverberation suppressor  53 , the signal delays in accordance with the order of the adaptive filter, as is apparent from  FIG. 5 . To compensate for this, the audio processing apparatus in  FIG. 1  includes the first delay device  55  and the second delay device  85 . Typically, a delay ½ (=M/2) the filter order of the reverberation suppressor  53  is given (when M is an odd number, a neighboring value is usable). At this time, for example, h(M/2)=1 is set, and all the other values h are initialized to 0. This allows the adaptive algorithm to run using the initial value in the no reverberation state. If an appropriate initial value for reverberation suppression is stored in the memory, the operation may be started after initializing h to that value. For example, the initial value may be set in the following way. The filter coefficient can be estimated to some extent based on the design values such as the dimensions around the microphones  7   a  and  7   b  and the material of the structure. Hence, the filter coefficient obtained from the design values may be set as the initial value. Alternatively, the filter coefficient when the audio recorder has been powered off may be stored in the memory and set as the initial value when activating the audio recorder next time. Otherwise, the filter coefficient may be calculated by generating predetermined reference sound in the production process of the audio recorder and stored in the memory, and used as the initial value when activating the audio recorder. 
     The operation of the ALC  61  will be described next. The ALC is provided to effectively utilize the dynamic range while suppressing saturation of the audio signal. Since the audio signal exhibits a large power variation on the time base, the level needs to be appropriately controlled. The level controller  63  provided in the ALC  61  monitors the outputs from the variable gains  62   a  and  62   b.    
     The attack operation will be explained first. Upon determining that the signal of higher level has exceeded a predetermined level, the gain is reduced by a predetermined step. This operation is repeated at a predetermined period. This operation is called the attack operation. The attack operation enables to prevent saturation. 
     The recovery operation will be described next. If the signal of higher level does not exceed a predetermined level for a predetermined time, the gain is increased by a predetermined step. This operation is repeated at a predetermined period. This operation is called the recovery operation. The recovery operation enables to obtain sound in a silent environment. 
     The variable gains  62   a  and  62   b  in the ALC  61  operate synchronously. Thus, when the gain of the variable gain  62   a  decreases by the attack operation, the gain of the variable gain  62   b  also decreases as much. With this operation, the level difference between the signal channels is eliminated, and the sense of incongruity decreases when the signals of the channels are mixed by the mixer  71 . 
     The wind-detector  81  will be described next. Let w 1  be wind noise picked up by the first microphone  7   a , and w 2  be wind noise picked up by the second microphone  7   b . The BPFs  82   a  and  82   b  do not mask the wind noise because the power of wind noise concentrates to the lower frequency range, as described above with reference to  FIGS. 3A to 3F . Thus, (w 1 −w 2 ) representing the level difference between the output signal of the first microphone  7   a  and the output signal of the second microphone  7   b  is obtained as the output of the subtracter  83 . Note that the above-described influence of reverberation is assumed to be negligible. In an actual environment as well, the influence of reverberation is negligible because it is much smaller than the wind noise. 
     The level detector  86  performs absolute value calculation of the output of the subtracter  83  and then appropriately performs LPF processing. The cutoff frequency of the LPF is determined based on the stability and detection speed of the wind-detector, and about 0.5 Hz suffices. The LPF operates to integrate a signal in the masking range and directly pass a signal in the passband. As a result, the same effect as that of integration operation+HPF can be obtained. Thus, the output becomes large when the absolute value calculation maintains high level for a predetermined time (the time changes depending on the above-described cutoff frequency). Thus, this is equivalent to monitoring Σ|w 1 −w 2 | for an appropriate time. 
       FIGS. 6A to 6D  show examples of the output signal of the wind-detector  81  which changes depending on the wind strength.  FIGS. 6A ,  6 B, and  6 C are views showing signals obtained by the first microphone  7   a  and the second microphone  7   b . The abscissa represents time, and the ordinate represents the signal level. Referring to  FIGS. 6A ,  6 B, and  6 C, the signal level +1 indicates the level at which a signal in the positive direction is saturated.  FIG. 6A  shows the signal in the windless state,  FIG. 6B  shows the signal when the wind is weak, and  FIG. 6C  shows the signal when the wind is strong. As is apparent, as the wind strength increases, the signal level of the first microphone  7   a  rises, and wind noise is generated. On the other hand, the signal level of the second microphone  7   b  does not so largely increase as compared to that of the first microphone  7   a , as can be seen. This indicates that the wind noise is reduced by the effect of the audio resistor  41 . 
       FIG. 6D  shows a result obtained by the above-described processing of the wind-detector  81 . In  FIG. 6D , the abscissa represents time, like  FIGS. 6A ,  6 B, and  6 C, and the ordinate represents the output of the wind-detector. Note that the passband of the BPFs  82   a  and  82   b  is 30 Hz to 1 kHz, and the cutoff frequency of the LPF in the level detector  86  is 0.5 Hz. As is apparent, the output of the wind-detector  81  remains almost zero in the windless state and increases its value as the wind becomes stronger. In  FIG. 6D , the signal near 0 sec is small because rising delays due to the influence of the LPF in the level detector  86 . Until wind detection, a delay as illustrated occurs in the leading edge of the signal in  FIG. 6D . When the delay is made smaller, the wind-detector is readily affected by fluctuations of wind. In this embodiment, wind detection is done with a delay as shown in  FIG. 6D . 
     The output of the wind-detector  81  is used for the switch  87  of the above-described reverberation suppressor  53  and also used to switch the HPF  52  to be described later and switch the mixing processing in the mixer  71 . 
     The operation of the mixer  71  will be described next with reference to  FIGS. 7A to 7D . Changing the variable gain  74  and the cutoff frequency of the HPF  73  based on the output of the wind-detector  81  has been described with reference to  FIG. 1 . A detailed changing method will be described with reference to  FIGS. 7A to 7D . 
       FIGS. 7A and 7C  show examples of the arrangement of the mixer  71 .  FIGS. 7B and 7D  are graphs showing methods of changing the variable parts in  FIGS. 7A and 7C , respectively. 
     The arrangement shown in  FIG. 7A  will be described. The mixer  71  shown in  FIG. 7A  has the same arrangement as that in  FIG. 1 . Referring to  FIG. 7A , the cutoff frequency (first cutoff frequency) of the HPF  73  is variable, whereas the cutoff frequency (second cutoff frequency) of the LPF  72  is fixed to, for example, 1 kHz. The upper graph of  FIG. 7B  schematically represents the gain of the variable gain  74 , and the lower graph schematically represents the cutoff frequency of the HPF  73 . The abscissa of  FIG. 7B  is common to the two graphs. Wn 1 , Wn 2 , and Wn 3  are thresholds representing the level of wind noise and indicate that the wind noise becomes stronger in this order. 
     As shown in  FIG. 7B , when the wind noise is smaller than the first threshold Wn 1 , wind processing is unnecessary. Hence, the gain of the variable gain  74  is set to the first lower limit value (for example, 0), and the cutoff frequency of the HPF  73  is set to the second lower limit value (for example, 50 Hz). As a result, the signal from the second microphone  7   b  is completely masked via the circuit shown in  FIG. 7A , and the signal in the audible range (where frequencies higher than the cutoff frequency of the HPF  73 , that is, 50 Hz, are the dominant components of sound) can be obtained only from the first microphone  7   a . Since the signal of the second microphone  7   b  provided with the audio resistor  41  need not be used, the object sound is supposedly obtained faithfully. 
     A case will be described in which the wind noise level falls within the range from the first threshold Wn 1  (inclusive) to the second threshold Wn 2  (exclusive). Within this range, as the wind noise level rises, the variable gain  74  is increased, and the cutoff frequency of the HPF  73  is raised. This control is performed to gradually increase, in the low-frequency audio signal, the ratio of the signal from the second microphone  7   b  provided with the audio resistor  41 . The wind noise largely acts on the signal from the first microphone  7   a . However, the wind noise is reduced by raising the cutoff frequency of the HPF  73 . 
     A case will be described in which the wind noise level falls within the range from the second threshold Wn 2  (inclusive) to the third threshold Wn 3  (exclusive). At this time, the value of the variable gain  74  is fixed to a predetermined upper limit value (for example, 1), and the cutoff frequency of the HPF  73  is raised as the wind noise level rises. Performing this control allows to further reduce the wind noise, although the audio that exists from the cutoff frequency of the LPF  72  to the cutoff frequency of the HPF  73  is lost. The cutoff frequency of the HPF  73  is not raised beyond an appropriate value because if it excessively rises, the object sound degrades too much. In the example of  FIG. 7B , when the wind noise level is equal to or more than the third threshold Wn 3 , the cutoff frequency of the HPF  73  is fixed to 2 kHz and does not change any more. 
     The arrangement shown in  FIG. 7C  that is another example will be described. The mixer  71  shown in  FIG. 7C  includes a variable LPF  76  in place of the fixed LPF  72  and the variable gain  74 . The upper graph of  FIG. 7D  schematically represents the cutoff frequency of the variable LPF  76 , and the lower graph schematically represents the cutoff frequency of the HPF  73 . The abscissa of  FIG. 7D  is common to the two graphs. Wn 1 , Wn 2 , and Wn 3  are thresholds representing the level of wind noise and indicate that the wind noise becomes stronger in this order. 
     As shown in  FIG. 7D , when the wind noise level is smaller than the first threshold Wn 1 , wind processing is unnecessary. Hence, the cutoff frequencies of the variable LPF  76  and the HPF  73  are set to 50 Hz. As a result, the signal from the second microphone  7   b  is almost completely masked via the circuit shown in  FIG. 7C , and the signal in the audible range (where frequencies higher than the cutoff frequency of the HPF  73 , that is, 50 Hz, are the dominant components of sound) can be obtained only from the first microphone  7   a . Since the signal of the second microphone  7   b  provided with the audio resistor  41  need not be used, the object sound is supposedly obtained faithfully. 
     A case will be described in which the wind noise level falls within the range from the first threshold Wn 1  (inclusive) to the second threshold Wn 2  (exclusive). Within this range, as the wind noise level rises, the cutoff frequencies of the variable LPF  76  and the HPF  73  rise while, for example, remaining identical. This control is performed to gradually use the signal from the second microphone  7   b  provided with the audio resistor  41  as the low-frequency audio signal. The wind noise largely acts on the signal from the first microphone  7   a . However, the wind noise is reduced by raising the cutoff frequency of the HPF  73 . 
     A case will be described in which the wind noise level falls within the range from the second threshold Wn 2  (inclusive) to the third threshold Wn 3  (exclusive). At this time, the cutoff frequency of the variable LPF  76  is fixed to a predetermined value (for example, 1 kHz), whereas the cutoff frequency of the HPF  73  is raised as the wind noise level rises. This control is performed to further reduce the wind noise, although the audio that exists from the cutoff frequency of the variable LPF  76  to the cutoff frequency of the HPF  73  is lost. The cutoff frequency of the HPF  73  is not raised beyond an appropriate value because if it excessively rises, the object sound degrades too much. In the example of  FIG. 7D , when the wind noise level is equal to or more than the third threshold Wn 3 , the cutoff frequency of the HPF  73  is fixed to 2 kHz and does not change any more. 
     An example has been described above in which the HPF  73  is operated in a range wider than that of the operations of the variable gain  74  and the variable LPF  76 . The HPF  73  may be operated only in the same range as that of the operations of the variable gain  74  and the variable LPF  76  by setting Wn 2 =Wn 3  obviously. When the operation is limited, the object sound can faithfully be acquired, although the wind noise reduction effect becomes small. On the other hand, the level of the wind noise generated in the first microphone  7   a  when the wind blows largely changes depending on the attachment structure of the microphone or the like. Settings of Wn 1 , Wn 2 , and Wn 3  are adjusted by comparing, for example, the necessity of wind noise reduction with the necessity of faithfully acquiring an object sound. 
     The range where the cutoff frequency of the variable LPF or LPF changes in the example of the mixer  71  shown in  FIGS. 7A to 7D  has been described above in detail. Examples of the cutoff frequency changeable range and the filter arrangement will briefly be described. 
     The mixer  71  of this embodiment mixes audios acquired by the plurality of microphones  7   a  and  7   b . In the processing of mixing signals of separated bands, particularly, the signals of the plurality of microphones preferably have the same phase on the respective paths in the overlapping frequency band. If the phases are shifted by the processing in the plurality of paths, the waveforms may cancel each other because they do not accurately match. To sufficiently meet this requirement, the HPF  73  and the LPF  72  are preferably formed from FIR filters of the same order. Using the FIR filters makes it possible to consistently mix the signals even when a so-called group delay properly is obtained, and processing is performed for each band. If the cutoff frequency of the FIR filter is very low (exactly speaking, if the ratio is very low when standardizing by the ratio to the sampling frequency), a filter of a very high order is necessary for obtaining sufficient filter performance. This is derived from the fact that a number of samples are required to obtain the wave of the frequency of the masking/passing target. Since the order of the filter cannot be increased infinitely, the lower limit of the cutoff frequency changeable range is determined. In the arrangement shown in  FIG. 7C , the LFP and the HPF are variable. Hence, the order of the variable LPF  76  and the HPF  73  becomes very high if the cutoff frequency is very low. Thus, in the examples shown in  FIGS. 7A to 7D , the lower limit of the frequency is set to 50 Hz not to largely affect the signal in the audible range. As described above, the frequency is not limited to 50 Hz and can appropriately be set in accordance with the computational resource. In the example shown in  FIG. 7A , only the HPF is variable. Hence, only one filter of high order as described above suffices. This arrangement has an advantage over that in  FIG. 7C  in terms of calculation amount reduction. 
     On the other hand, the upper limit of the changeable range is determined by the second microphone  7   b  provided with the audio resistor  41 . As schematically shown in  FIG. 3B , the band of the object the second microphone  7   b  can acquire is limited to f 0  by the influence of the audio resistor  41 . Beyond this, no object sound is obtained. Hence, in the examples shown in  FIGS. 7A to 7D , the cutoff frequencies of the variable LPF  76  and the HPF  73  should be set lower. In  FIGS. 3A to 3F , the frequency is f 1 , and it should obviously satisfy f 1 &lt;f 0 . 
     The effect and variable operation of the HPF  52  will be described with reference to  FIGS. 1 ,  3 A to  3 F,  6 A to  6 D, and  8 A to  8 D to  11 A and  11 B. As described above with reference to  FIGS. 3A to 3F  and  6 A to  6 D, the wind noise concentrates to the lower frequency range and affects the first microphone  7   a  and the second microphone  7   b  in much different ways. Thus, even weak wind generates large wind noise in the first microphone  7   a . Problems caused by this are saturation of the ADC  54   a  and an inappropriate operation of the ALC  61 . Saturation of the ADC  54   a  is easily understandable, and a description thereof will be omitted. The problem of the operation of the ALC  61  at the time of wind noise generation will be explained. 
     If the HPF  52  does not exist, large wind noise is generated in the first microphone  7   a , as shown in  FIGS. 6A to 6D . Even if the wind noise and the object sound are superposed, the wind noise is assumed to be dominant. In such an environment, the ALC  61  performs level control by referring to the wind noise level of the first microphone  7   a . When the HPF  73  in the mixer  71  then processes the wind noise, the level of the audio signal greatly lowers. As a result, the output of the adder  75  is very small. Thus, the signal level is inappropriate. 
     To solve the above-described problems such as the saturation of the ADC and the inappropriate signal level, for example, the technique of patent literature 1 may be applied. However, according to the related art, the circuit scale becomes large because the ALC operation is performed at two portions, and the quantization error may also increase. 
     Consider the HPF  52  shown in  FIG. 1 , which is the second high-pass filter for suppressing wind noise. When the cutoff frequency (third cutoff frequency) of the HPF  52  is appropriately set, the main components of wind noise can be removed. This enables to prevent saturation of the ADC  54   a  and allows the ALC  61  to appropriately control the gain (since the object sound is not buried in wind noise at the point of the ALC  61 , an ALC operation corresponding to the level of the object sound can be performed). 
     An example of the cutoff frequency control sequence of the HPF  52  will be described with reference to  FIGS. 8A to 8D .  FIG. 8A  shows the operation sequence of the switch  87 .  FIG. 8B  shows the operation sequence of the HPF  52 .  FIG. 8C  shows the operation sequence of the variable gain  74 .  FIG. 8D  shows the operation sequence of the HPF  73  that is the first high-pass filter for passing only the high-frequency components of the output signal of the ALC  61 . The abscissa representing the level of wind noise is common to  FIGS. 8A to 8D . Wn 1 , Wn 2 , and Wn 3  are thresholds representing the level of wind noise and indicate that the wind noise becomes stronger in this order. The operation in  FIGS. 8C and 8D  is the same as that in  FIG. 7B , and a description thereof will not be repeated. 
     When the wind noise level is smaller than the first threshold Wn 1 , wind processing is unnecessary. Hence, the switch  87  is turned on, and the adaptive operation of the reverberation suppressor  53  described above is performed. The cutoff frequency of the HPF  52  is set to 0 Hz (=through without the HPF operation). Since the signal of the second microphone  7   b  provided with the audio resistor  41  need not be used, the object sound is supposedly obtained faithfully. 
     When the wind noise level is equal to or more than the first threshold Wn 1 , wind noise is generated. Hence, the switch  87  is turned off, and the adaptive operation of the reverberation suppressor  53  described above is stopped. This control allows to suppress the inappropriate adaptive operation. 
     A case will be described in which the wind noise level falls within the range from the first threshold Wn 1  (inclusive) to the second threshold Wn 2  (exclusive). At this time, as the wind noise level rises, the cutoff frequency of the HPF  52  rises stepwise at a value lower than the cutoff frequency of the HPF  73 . Performing this control enables to reduce the wind noise generated in the first microphone  7   a . When the control is performed not to exceed the cutoff frequency of the HPF  73 , the cutoff frequency of the HPF  52  does not largely affect the output of the HPF  73 . 
     Effects obtained by this arrangement will be described. The HPF  52  is provided in the analog part (before the ADC) of the audio processing apparatus  51  and therefore formed from an IIR filter (an HPF formed from an RC circuit) in general. At this time, the HPF  52  cannot satisfy the group delay property. On the other hand, the phase delay is small in the passband even in the IIR filter. Thus, even if the group delay property is not satisfied, the phase delay does not affect. Controlling the cutoff frequencies of the HPFs  52  and  73  as described above makes it possible to reduce the influence of the phase delay caused by the IIR filter. As described above, in the processing of mixing signals of separated bands, particularly, the signals of the plurality of microphones preferably have the same phase on the respective paths in the overlapping frequency band. However, even if this condition is not satisfied, the influence can be reduced. In addition, the HPF  52  is provided in the analog part of the audio processing apparatus  51 . However, if the HPF  52  is configured to continuously change the cutoff frequency in the analog circuit, the circuit scale becomes large. When a circuit suitable for the control sequence described with reference to  FIGS. 8A to 8D  is formed, the HPF can be implemented by a simple arrangement. 
       FIGS. 9 and 10  show examples of signals processed by the above-described circuit.  FIG. 9  shows a case in which the HPF  52  is not provided.  FIG. 10  shows a case in which the HPF  52  is provided. The signals in  FIG. 9  are processed in a state in which the HPF  52  is removed from the arrangement in  FIG. 1 . As illustrated, the graphs represent the output of the gain  62   a , the output of the gain  62   b , the output of the HPF  73 , the output of the LPF  72 , and the output of the adder  75 , respectively, sequentially from the upper side. The abscissa represents time and is common to all graphs. The examples shown in  FIGS. 9 and 10  indicate that the object speaks from near 2.5 sec (human voice is the sound to be collected). The signals shown in  FIGS. 9 and 10  are processed assuming that the wind noise level is Wn 2  in  FIGS. 8A to 8D . 
     Only wind noise exists before 2.5 sec, as in the graphs of  FIGS. 6A to 6D . Placing focus only on this portion, the output of the gain  62   a  appears to be larger in  FIG. 10  than in  FIG. 9 . This is because the gain is actually increased by the ALC  61 . This is apparent from the portion after 2.5 sec where the output is superposed on the object sound. 
     Placing focus on the output of the gain  62   b  after 2.5 sec reveals that the signal in  FIG. 9  obviously has a signal level lower than that of the signal in  FIG. 10 . This is because the gain becomes smaller because of the level control performed by the ALC  61  for the wind noise generated in the first microphone  7   a , and the object sound is consequently acquired very small. On the other hand, in the signal shown in  FIG. 10 , the wind noise generated in the first microphone  7   a  is reduced by the effect of the HPF  52 , and the gain of the ALC  61  is kept high as compared to the state of  FIG. 9 . 
     Placing focus on the output of the HPF  73  in  FIG. 9  reveals that the wind noise is considerably reduced by appropriately processing the cutoff frequency of the HPF  73 . However, since the signal level of the output of the HPF  73  is much lower than that of the output of the gain  62   a , the signal level of the final output from the adder  75  is very low, as can be seen. 
     On the other hand, even in  FIG. 10 , the wind noise is considerably reduced by appropriately processing the cutoff frequency of the HPF  73 , as is apparent. In addition, since the output of the LPF  72  remains large, the signal level of the final output from the adder  75  is also kept at a sufficient level, as can be seen. 
     As described above, when the HPF  52  is arranged on a side closer to the microphone than the ADC and the ALC, a high-quality audio can be obtained. 
       FIGS. 11A and 11B  illustrate other examples of the circuit arrangement of this embodiment.  FIG. 11A  shows an example in which the ALC is arranged in the analog part.  FIG. 11B  shows an example in which the ALC  61  is arranged after the mixer  71 . Even such an arrangement enables to obtain the effects described in this embodiment. 
     As described above, according to this embodiment, it is possible to obtain a high-quality audio with suppressed wind noise by a simple circuit arrangement. 
     Second Embodiment 
     An audio recorder and an image capture device including the audio recorder according to the second embodiment of the present invention will be described below with reference to  FIGS. 12 and 13 . The same reference numerals as in the first embodiment denote parts that perform the same operations in the second embodiment. 
       FIG. 12  is a perspective view showing the image capture device. Although the apparatus in  FIG. 12  is similar to that of  FIG. 2A , an opening portion  32   c  for a microphone is added. A microphone  7   c  (not shown) is provided behind the opening portion  32   c.    
       FIG. 13  is a block diagram for explaining the main part of an audio processing apparatus  51  corresponding to the apparatus shown in  FIG. 12 . In  FIG. 13 , the arrangement is extended to a stereo system based on the circuit including the ALC in the analog part according to the first embodiment shown in  FIG. 11A . The illustrations of a reverberation suppressor  53  and a level detector  86  are simplified/changed. A first microphone  7   a  is extended to two microphones, unlike the first embodiment. The microphones  7   a  and  7   c  respectively constitute the left and right channels of the stereo system and are designed to have the same characteristic. On the other hand, a second microphone  7   b  is provided with an audio resistor  41  and has the same characteristic as in the first embodiment. 
     An HPF  52   b , a gain  62   c , an ADC  54   c , a DC component cutting HPF  56   c , and an HPF  73   b  extended in  FIG. 13  perform the same operations as those of the HPF  52 , the gain  62   a , the ADC  54   a , the DC component cutting HPF  56   a , and the HPF  73  described in the first embodiment, respectively. Delay devices  55   a  and  55   b , a newly provided phase comparator  57 , an adder  58 , and a gain  59  whose operations change will be described here. 
     In the stereo audio recorder, the signal are given the stereo effect by the phase difference between the audio signals. In the arrangement shown in  FIG. 12 , the second microphone  7   b  is arranged between the first microphones  7   a  and  7   c . In this arrangement, when the phase difference between the microphones  7   a  and  7   c  is considered, the phase of the signal of the second microphone  7   b  exists between them. For example, when the second microphone  7   b  is arranged just at the intermediate point equidistant from the microphones  7   a  and  7   c , the phase also exists at the intermediate point. In the circuit shown in  FIG. 13 , the phase difference between the microphones  7   a  and  7   c  is calculated, and a delay corresponding to it is given by the delay devices  55   a  and  55   b.    
     For example, examine a case in which the signal of the microphone  7   c  delays from that of the microphone  7   a . At this time, the reverberation suppressor is controlled to comply with the intermediate signal, as will be described later. When mixing with the signal of the microphone  7   a , the phase is advanced. When mixing with the signal of the microphone  7   c , the phase is delayed to mix the signals. In the first embodiment, a delay ½ (=M/2) the filter order of the reverberation suppressor  53  is given. The delay device  55   a  gives a smaller delay, and the delay device  55   b  gives a larger delay. The absolute value changes depending on the position of the microphone. For example, when the second microphone  7   b  is located at the intermediate point between the first microphones  7   a  and  7   c , as described above, each phase is shifted by ½ the phase difference calculated by the phase comparator  57 . Performing the above-described processing allows to obtain an audio signal without reducing the stereo effect. 
     The adder  58  and the gain  59  will be explained. The adder  58  adds the signals of the microphones  7   a  and  7   c . The gain  59  halves the output of the adder  58 . As a result, the output of the gain  59  is the average of the microphones  7   a  and  7   c . A thus obtained audio signal has the intermediate phase between the signals of the microphones  7   a  and  7   c . On the other hand, a BPF  82   a  passes only a band of about 30 Hz to 1 kHz, as described above in the first embodiment. The audio processing apparatus  51  is configured to acquire even an audio signal of a frequency higher than the passband of the BPF. As for the audio signal acquirable at this time, the microphones  7   a  and  7   c  are arranged such that no phase inversion occurs between their signals. When observing only in the passband of the BPF  82   a , the phase difference between the signals of the microphones  7   a  and  7   c  is small. Hence, the levels of the signals in the passband of the BPF  82   a  can be considered to be almost added. Thus, when the gain  59  halves the output, a signal having a signal level almost equal to that of the first microphones  7   a  and  7   c  and a phase at the intermediate point can be obtained. In this embodiment, the reverberation suppressor  53  is operated so as to comply with the output of the gain  59  described above. 
     With the above-described arrangement, the present invention is easily applicable even to a stereo audio recorder without reducing the stereo effect. 
     In this embodiment, a stereo apparatus (including two first microphones for acquiring a high-frequency range) has been described. The arrangement can easily be extended to an audio recorder including more microphones. 
     Other Embodiments 
     Aspects of the present invention can also be realized by a computer of a system or apparatus (or devices such as a CPU or MPU) that reads out and executes a program recorded on a memory device to perform the functions of the above-described embodiments, and by a method, the steps of which are performed by a computer of a system or apparatus by, for example, reading out and executing a program recorded on a memory device to perform the functions of the above-described embodiments. For this purpose, the program is provided to the computer for example via a network or from a recording medium of various types serving as the memory device (for example, computer-readable medium). 
     While the present invention has been described with reference to exemplary embodiments, it is to be understood that the invention is not limited to the disclosed exemplary embodiments. The scope of the following claims is to be accorded the broadest interpretation so as to encompass all such modifications and equivalent structures and functions. 
     This application claims the benefit of Japanese Patent Application No. 2011-027843 Feb. 10, 2011, which is hereby incorporated by reference herein in its entirety.