Patent Publication Number: US-8976978-B2

Title: Sound signal processing apparatus and sound signal processing method

Description:
BACKGROUND 
     The present disclosure relates to a sound signal processing apparatus and a sound signal processing method for obtaining a sound from a specific sound source direction. 
     Japanese Unexamined Patent Application Publication No. 2010-11117 and Japanese Unexamined Patent Application Publication No. 2007-129383 are examples of related art. 
     For example, a beam forming technique for forming directivity with respect to input sounds from two microphones is known. 
       FIG. 10  illustrates an example of a noise cancellation headphone (hereinafter, referred to as an NC headphone). Although an NC headphone  100  supplies stereo sounds to a user using left and right speakers  101 L and  101 R, microphones  102 L and  102 R that absorb external sounds are provided in order to reduce external noises. 
     The NC headphone  100 , for example, reproduces and outputs the sounds of reproduction music from a portable media player or the like. 
     In brief, in order to cancel noises, reversed phase components of sound signals absorbed by the microphones  102 L and  102 R are generated, combined with respective music signals, and then output from the speakers  101 L and  101 R. Therefore, the sounds of the music signals are listened to by a user in a state in which external noises are spatially cancelled. 
     Here, it is considered that the microphones  102 L and  102 R are used not only for canceling noise but also for absorbing external sounds which have directivity. 
     For example, although it is preferable that a user can normally perform conversation or the like even when wearing the NC headphone  100 , if a noise cancellation function is turned on, for example, even the sounds of a person who is in front of the user are reduced, so that it is difficult to listen to conversational sounds. 
     Therefore, for example, when a conversation or the like is performed, a mode that turns off reproduction music and turns off a noise cancellation function is provided. 
     However, if the noise cancellation function is turned off, surrounding noise is heard to a large degree together with the sounds of other people. Therefore, in a place where there is much traffic, the inside of a plane, or the like, a state in which conversational sounds or the like are difficult to be heard is not changed. 
     In such a case, it is preferable that a speaker output, in which conversational sounds are easily heard and surrounding noises are suppressed, can be realized. 
     If it is considered that a user wears the NC headphone  100  and faces the front as in  FIG. 10 , it can be considered that the sounds of a target that conducts conversation come from the front of the user in most cases. At this time, as shown in  FIG. 10 , when viewed from the user, the user regards sound sources other than from the front as noises, of which the level thereof should be lowered while boosting the conversational sounds from the front. 
     In order to realize this, when a necessary sound source direction is temporarily set to the front, directivity can be formed at the time of absorbing sounds using a so-called beam forming method. 
       FIG. 11A  is a conceptual diagram illustrating a beam forming process, and sound signals from left and right microphones  102 L and  102 R are processed and output by a beam forming process unit  103 . 
     The simplest beam forming process may be a process of adding sound signals from the left and right microphones  102 L and  102 R as shown in  FIG. 11B  when the necessary directivity is the front or the back. 
     In this case, the phases of the sound signal components of left and right channels with respect to sounds from the front or back, that is, sounds from the sound sources at an equal distance from the microphones  102 L and  102 R, are matched with each other, and boosted by addition. Since the phases of the sound signal components of sounds from other directions are deviated from the phases of the sound signal components of the left and right channels, the sound signal components are reduced by as much as the deviation. Therefore, sound signals having, for example, directivity in the front direction can be obtained. 
     Meanwhile, the beam forming process itself can boost the sound signals in the directions other than the front direction. In this case, a delay unit is installed on one side channel, with the result that the time difference between the same wave fronts which reach each of the microphones can be absorbed, so that the beam forming can be realized in an oblique direction or in a traverse direction. 
     In order to increase the accuracy of the beam forming (in this case, the same meaning as the boosting of the front directivity and the reduction in surrounding noises), a noise suppression device that mainly uses band pass filters shown in  FIG. 12  is generally used and not the simple device as shown in  FIG. 11B . 
     The sound signal obtained by the microphone  102 L is amplified by the microphone amplifier  104 L, and then supplied to band pass filters  121 L,  122 L, and  123 L that have central pass frequencies fc 1 , fc 2 , and fc 3 , respectively. In the band pass filters  121 L,  122 L, and  123 L, the sound signal components of the bands BD 1 , BD 2 , and BD 3  are extracted. 
     Further, the sound signal obtained by the microphone  102 R is amplified by the microphone amplifier  104 R, and then supplied to band pass filters  121 R,  122 R, and  123 R that have central pass frequencies fc 1 , fc 2 , and fc 3 , respectively, so that the sound signal components of the respective bands BD 1 , BD 2 , and BD 3  are extracted. 
     Meanwhile, the pass band of the band pass filter that has the central frequency fc 1  is represented as a band BD 1 . In the same manner, the pass bands of the band pass filters that have the central frequencies fc 2  and fc 3  are represented as bands BD 2  and BD 3 . 
     The sound signal components of the band BD 1 , which are the outputs of the band pass filters  121 L and  121 R, are supplied to the sound source directional angle analysis unit  124  and the adder  127 . 
     The sound signal components of the band BD 2 , which are the output of the band pass filters  122 L and  122 R, are supplied to the sound source directional angle analysis unit  125  and the adder  128 . 
     The sound signal components of the band BD 3 , which are the outputs of the band pass filters  123 L and  123 R, are supplied to the sound source directional angle analysis unit  126  and the adder  129 . 
     The sound source directional angle analysis units  124 ,  125 , and  126  determine the sound source direction of a dominant sound from among the sound signal components of the bands BD 1 , BD 2 , and BD 3 , respectively. 
     Thereafter, the sound source directional angle analysis units  124 ,  125 , and  126  control the gain of the variable gain amplifiers  130 ,  131 , and  132  based on the determined direction. That is, the sound source directional angle analysis units  124 ,  125 , and  126  perform control such that the gain increases when the determined direction is a target direction, such as the front direction or the like, and such that the gain decrease when the determined direction is the other direction. 
     Each of the sound signal components of the bands BD 1 , BD 2 , and BD 3  is added by the adders  127 ,  123 , and  129  for the respective L and R channels, and then supplied to the variable gain amplifiers  130 ,  131 , and  132 . Thereafter, the variable gain amplifiers  130 ,  131 , and  132  are controlled by the sound source directional angle analysis units  124 ,  125 , and  126  as described above, so that, for example, a band in which the sound from the front direction is dominant is boosted, and the other bands are reduced. The outputs of the respective bands BD 1 , BD 2 , and BD 3  in which gains are adjusted as weights for respective bands are added by an adder  133 , and become an output sound signal Sout on which a beam forming process has been performed. 
     When a beam forming process unit  103  using such a noise suppression device is used, the conversation sounds are not easily buried in noises and can be heard in the state as shown in  FIG. 10 . 
     Further, as one type of method of boosting sounds and suppressing noises, a method using an FFT that centers on “spectrum subtraction” may be provided as a representative method of analyzing and combining sounds without using the beam forming method in order to remove noises in the related art. 
     SUMMARY 
     As described above, as the representative method of analyzing and combining sounds in order to reduce noises in the related art, two methods that use band pass filters and FFTs respectively are provided. 
     The method using FFTs has some disadvantages. The first is that the calculation amount is enormous, and the second is that peculiar noise sounds which cause uncomfortable feeling termed musical noises are generated. 
     On the other hand, in the method using band pass filters as shown in  FIG. 12 , the calculation amount can be suppressed to be small and musical noises are not generated in principle. Further, there is an advantage in that variation in the quality and quantity of the process can be processed without making large changes. 
     As one reason behind this, since FFT can treat only the sample number of 2 n , for example, the calculation amount is discrete and is not increased by a small amount because there is a calculation resource. On the other hand, with respect to band pass filters, since a single band pass filter has a small unit of calculation amount, there are advantages in that the number of bands is easily increased and decreased and can be set in detail according to the calculation resource. Therefore, it is considered that the method using band pass filters is preferable. 
     However, in the method using band pass filters, there is a problem in that sound quality is decreased compared with before the process is performed. 
     When sounds are generally analyzed and combined using band pass filters, a method of analyzing the sound data of each of the bands divided by band pass filters, performing a process on the sound data of each of the bands in parallel, and finally combining all the sound data is used. 
     In a method of analyzing and combining sounds using band pass filters as shown in  FIG. 12 , sound quality is better than the case of FFTs. However, phase rotation is controlled and adjusted according to band pass filters, addition/non-addition is controlled and adjusted according to bands, or increase/decrease in a level is controlled and adjusted. Therefore, when addition is performed for the respective bands, phases may not be matched compared to the original sound source, so that sound quality deterioration felt as noises is undeniable and becomes a problem. 
     It is desirable to provide a signal processing method (noise suppression method based on beam forming) for reducing noises while maintaining sound quality with respect to sound signals obtained from a plurality of microphones, thereby improving calculation processing efficiency. 
     A sound signal processing apparatus according to an embodiment of the disclosure includes a sound source direction determination unit that determines sound source directions with respect to sound signals of a plurality of channels, which can be obtained by, for example, a plurality of microphone inputs or line inputs, for respective first to n-th bands; and a filter processing unit that includes first to n-th filters which are connected in series and configured to boost or attenuate the sound signals with respect to the first to n-th bands. The respective first to n-th filters perform boosting or attenuation based on the sound source directions of the first to n-th bands which are determined by the sound source direction determination unit. 
     Further, the sound source direction determination unit may include first to n-th sound source directional angle analysis units corresponding to the first to n-th bands. Each of the first to n-th sound source directional angle analysis units may have one-to-one correspondence with each of the first to n-th filters and regards the corresponding filters as control targets for a boosting or attenuating process. Each of the first to n-th sound source directional angle analysis units may allow the filter to be controlled to perform the boosting process when a sound source direction of a corresponding band is determined as a direction included in a predetermined angle range, and allow the filter to be controlled to perform the attenuating process when a sound source direction angle of the corresponding band is not determined as a direction included in the predetermined angle range. 
     Further, each of the first to n-th sound source directional angle analysis units may allow the filter to be controlled to perform the attenuating process when the sound source direction is determined be in a dispersion state. 
     Further, each of the first to n-th sound source directional angle analysis units may determine the sound source direction with respect to the corresponding band based on energy subtraction of the sound signals of the respective channels. 
     Further, each of the first to n-th filters of the filter processing unit, which are connected in series, may receive a sound signal with which the sound signals of the plurality of channels are combined. 
     Further, each of the first to n-th filters, which are connected in series, of the filter processing unit may receive a sound signal of one of the plurality of channels. 
     A sound signal processing method according to another embodiment of the disclosure may include determining sound source directions with respect to sound signals of a plurality of channels for respective first to n-th bands; and inputting sound signals to first to n-th filters which are connected in series and configured to boost or attenuate the sound signals with respect to the first to n-th bands, and performing boosting or attenuation by the respective first to n-th filters based on the sound source directions of the first to n-th bands, which are determined in the determining of the sound source directions. 
     The above-described disclosure is a signal processing method (a noise suppression method based on beam forming) for reducing noises while maintaining sound qualities of the apparatus which use two or more microphones, thereby improving calculation processing efficiency. 
     In order to remedy the deterioration in sound qualities accompanied by a noise reduction method, sound signals obtained by a single or two or more separated microphones are divided for a respective plurality of bands, and analysis (sound source direction determination) is performed to determine noises for the respective bands. 
     Thereafter, one or a plurality of additional values of the input sound signals are processed based on the analysis results of the sound source direction using a group of filters arranged in series on a time axis on which the mismatch of phases does not occur, thereby reducing noises. 
     The group of filters connected in series includes a plurality of band boosting or attenuation filters capable of controlling gains, and the filters are controlled based on the analysis results. 
     According to the disclosure, the sound signal process of reducing noises while maintaining sound qualities with respect to sound signals obtained from a plurality of microphones, thereby improving calculation processing efficiency, can be realized. 
    
    
     
       BRIEF DESCRIPTION OF THE DRAWINGS 
         FIG. 1  is a block diagram of a noise suppression device according to an embodiment of the disclosure; 
         FIG. 2  is an explanatory view of MPF characteristics according to the embodiment; 
         FIG. 3  is an explanatory view of sample plots at the time of sound source direction determination according to the embodiment; 
         FIGS. 4A to 4C  are explanatory views of the sound source direction determination according to the embodiment; 
         FIG. 5  is an explanatory view of MPF control based on the sound source direction determination according to the embodiment; 
         FIG. 6  is a flowchart of a process performed by a sound source directional angle analysis unit according to the embodiment; 
         FIG. 7  is an explanatory view of an example applied to an NC headphone according to the embodiment; 
         FIG. 8  is a block diagram of the NC headphone according to the embodiment; 
         FIG. 9  is a block diagram of a noise suppression device according to another embodiment; 
         FIG. 10  is an explanatory view of a conversation state in noisy conditions; 
         FIGS. 11A and 11B  are explanatory views of a beam forming process; and 
         FIG. 12  is a block diagram of a noise suppression device according to the related art. 
     
    
    
     DETAILED DESCRIPTION OF EMBODIMENTS 
     Hereinafter, embodiments of the disclosure will be described in the following order. 
     1. Noise suppression device according to Embodiment 
     2. Example applied to NC headphone 
     3. Examples applied to Various Kinds of Apparatus and Modified Example 
     1. Noise Suppression Device According to Embodiment 
       FIG. 1  shows a noise suppression device  1  as an embodiment of a sound signal processing apparatus of the disclosure. 
     The noise suppression device  1  obtains sound signals which are preferable to a conversation in a noise environment in such a way that sound signals absorbed by the left and right microphones  2 L and  2 R are input, and sounds from, for example, the front (or the back), are boosted, and sounds from the other directions are attenuated. 
     In  FIG. 1 , a sound signal SmL obtained by the microphone  2 L is amplified by the microphone amplifier  3 L, and is converted to digital data by an Analog-to-Digital (A/D) converter  4 L. Thereafter, the sound signal SmL which is converted to the digital data is input to the noise suppression device  1 . 
     Further, a sound signal SmR obtained by the microphone  2 R is amplified by the microphone amplifier  3 R, and is converted to digital data by an A/D converter  4 R. Thereafter, the sound signal SmR which is converted to the digital data is input to the noise suppression device  1 . 
     The noise suppression device  1  is configured to include a sound source direction determination unit  1 A and a filter processing unit  1 B. 
     The sound source direction determination unit  1 A determines the sound source directions of the sound signals SmL and SmR of L and R channels for respective first to third bands in this example. 
     The filter processing unit  1 B includes first to third filters (MPFs  58 ,  59 , and  60  which will be described later) that are configured to boost or attenuate the sound signals for the respective first to third bands and are connected to each other in series. 
     The sound source direction determination unit  1 A includes band pass filters  51 L,  52 L,  53 L,  51 R,  52 R and  53 R, and sound source directional angle analysis units  54 ,  55 , and  56 . 
     The central pass frequencies of the respective band pass filters  51 L,  52 L, and  53 L are set to fc 1 , fc 2 , and fc 3 . For the explanation, the respective pass bands are represented as BD 1 , BD 2 , and BD 3 . 
     Further, the central pass frequencies of the respective band pass filters  51 R,  52 R, and  53 R are set to central pass frequencies fc 1 , fc 2 , and fc 3 . The respective pass bands are represented as BD 1 , BD 2 , and BD 3  in the same manner. 
     The sound signal SmL of the left channel is input to the band pass filters  51 L,  52 L, and  53 L, and the sound signal components of the respective bands BD 1 , BD 2 , and BD 3  are extracted. 
     Further, the sound signal SmR of the right channel is input to the band pass filters  51 R,  52 R, and  53 R, and the sound signal components of the respective bands BD 1 , BD 2 , and BD 3  are extracted. 
     The sound signal components of the band BD 1  of each of the left and right channels, which are the outputs of the band pass filters  51 L and  51 R, are provided to the sound source directional angle analysis unit  54 . 
     The sound signal components of the band BD 2  of each of the left and right channels, which are the outputs of the band pass filters  52 L and  52 R, are provided to the sound source directional angle analysis unit  55 . 
     The sound signal components of the band BD 3  of each of the left and right channels, which are the outputs of the band pass filters  53 L and  53 R, are provided to the sound source directional angle analysis unit  56 . 
     The sound source directional angle analysis unit  54  corresponds to the band BD 1 , and determines the sound source direction of a dominant sound from among the supplied sound signal components of the band BD 1 . 
     The sound source directional angle analysis unit  55  corresponds to the band BD 2 , and determines the sound source direction of a dominant sound from among the supplied sound signal components of the band BD 2 . 
     The sound source directional angle analysis unit  56  corresponds to the band BD 3 , and determines the sound source direction of a dominant sound from among the supplied sound signal components of the band BD 3 . 
     Although a method of determining the sound source directions by the sound source directional angle analysis units  54 ,  55 , and  56  will be described later, each of the sound source directional angle analysis units  54 ,  55 , and  56  determines the sound source direction based on the energy subtraction of the sound signals of the respective channels with respect to the corresponding bands. 
     Thereafter, the sound source directional angle analysis units  54 ,  55 , and  56  control Mid Presence Filters (MPFs)  58 ,  59 , and  60 , which are provided to has one-to-one correspondence with the sound source directional angle analysis units  54 ,  55 , and  56 , using control signals SG 1 , SG 2 , and SG 3  according to the determined directions. As can be understood from the drawing, an MPF  58  serves as a control target of the sound source directional angle analysis unit  54 , an MPF  59  serves as a control target of the sound source directional angle analysis unit  55 , and an MPF  60  serves as a control target of the sound source directional angle analysis unit  56 , respectively. 
     The filter processing unit  1 B includes an adder  57 , and MPFs  58 ,  59 , and  60 . The MPFs  58 ,  59 , and  60  serve as a series-connected filter group. 
     An adder  57  adds the sound signals SmL and SmR of the left and right channels. A sound signal (LR addition signal), in which the sound signals of the left and right channels are combined by the adder  57 , is supplied to the MPF  58 . 
     The MPFs  58 ,  59 , and  60  boost or attenuate the corresponding bands, respectively. Here, the reason why three MPFs are provided is that the band pass filters  51 L,  52 L,  53 L,  51 R,  52 R, and  53 R of the sound source direction determination unit  1 A divide each of the sound signals SmL and SmR into three bands. 
     The central pass frequencies of the respective MPFs  58 ,  59 , and  60  are set to fc 1 , fc 2 , and fc 3 . Further, each of the MPFs  58 ,  59 , and  60  has characteristics as shown in  FIG. 2 , and performs amplification and reduction of gain with respect to a specific target band (a band centering on a frequency fc). The boosting and attenuation of a target band attributable to such variable gain adjustment in the MPFs  58 ,  59 , and  60  are controlled by the sound source directional angle analysis units  54 ,  55 , and  56 , as described above. 
     That is, although the MPF  58  boosts and attenuates the band BD 1  which centers on the frequency fc 1 , the MPF  58  corresponds to the band pass filters  51 L and  51 R and the sound source directional angle analysis unit  54 . 
     Further, although the MPF  59  boosts and attenuates the band BD 2  which centers on the frequency fc 2 , the MPF  59  corresponds to the band pass filters  52 L and  52 R and the sound source directional angle analysis unit  55 . 
     Further, although the MPF  60  boosts and attenuates the band BD 3  which centers on the frequency fc 3 , the MPF  60  corresponds to the band pass filters  53 L and  53 R and the sound source directional angle analysis unit  56 . 
     Thereafter, when the noise suppression device  1  sets a front (back) direction as a target direction, a band in which a sound source direction is determined as the front (back) direction is boosted, and a band in which the sound source direction is determined as another direction is attenuated. A boosting/attenuation level is based on the determination of a directional angle. 
     In the respective MPFs  58 ,  59 , and  60 , the sound signal (LR addition signal) is boosted or attenuated based on the control by the sound source directional angle analysis units  54 ,  55 , and  56 . Thereafter, the output of the MPF  60  is the output signal Sout of the noise suppression device  1 . 
     The determination process of the sound source directional angle analysis units  54 ,  55 , and  56  and the control with respect to the MPFs  58 ,  59 ,  60  will be described. 
       FIG. 3  shows the plots of sample values at the time of the sound source direction/angle determination performed by the sound source directional angle analysis units  54 ,  55 , and  56 . 
     Although the components of the bands BD 1 , BD 2 , and BD 3  of the sound signals SmL and SmR are input to the respective sound source directional angle analysis units  54 ,  55 , and  56 , the sound source directional angle analysis units  54 ,  55 , and  56  plot the amplitude values of the respective L and R channels. 
     The plot locations on the LR plane of  FIG. 3  represent the energy subtraction of the sound signals SmL and SmR of the respective L and R channels. 
     First, the absolute values of the amplitude values of the L/R channels of a target band are plotted on the LR plane of  FIG. 3 , and this process is repeated during a specific time period. 
     For example, as an input value at a certain time point t 0 , if it is assumed that the absolute value of the amplitude of the L channel is set to A 1  and the absolute value of the amplitude of the R channel is set to A 2 , the input value is plotted as a sample SPt 0  represented as the black circle. This process is sequentially performed on each of the time points t 1 , t 2 , . . . , and samples SPt 1 , SPt 2 , . . . are plotted as shown in the drawing. 
     If a plurality of samples SPs are plotted during a certain unit time (for example, determined as about 0.5 to 5 seconds), a straight line LL starting from an original point is obtained using a least-squares method. That is, a straight line in which the sum of the squares of the distance from all the samples SPs becomes minimum, and the straight line is set to the straight line LL. 
     The angle θ of the straight line LL is regarded as the angle of the sound source direction. 
     With respect to the sound signal of a certain band, when an angle θ (straight line LL) comes around the center of the LR plane (around 45° in the drawing), it can be considered that the difference in the amplitude values in the corresponding band is small and it can be considered that the sound source is equidistant from the right and left. That is, the front direction can be estimated as the sound source direction. 
     On the other hand, when the angle θ (straight line LL) is inclined to the longitudinal axis or inclined to the lateral axis of the LR plane, it can be considered that the difference in the right and left amplitude values of the sound of the band is large and it can be considered that the sound from the right direction side or the left direction side. 
     Here, for example, as shown as oblique line sections in  FIGS. 4A to 4C , the area of the angle θ in which the straight line LL exists around 45° is regarded as a center area. The center area is an area in which the sound source direction is regarded as the front (or back). On the other hand, a right area and a left area in the drawing correspond to areas in which the sound source directions are regarded as the right side and the left side, respectively. In  FIGS. 4A to 4C , black circles are the plot points of the sample SPs. 
     For example, when the state described with reference to  FIG. 10  is considered, the sound source direction of a conversational sound from another person can be considered as a front direction. In this case, the sound signal component of bands in which the sound source direction is the front direction can be estimated as, for example, the conversational sound, that is, the sound which a user wants to hear. On the other hand, the sound signal component of bands in which the sound source direction is another direction can be estimated as a noise sound, that is, the sound signal component desired to be reduced. 
     In this case, when the angle θ exists within the range of the center area on the LR plane as shown in  FIG. 4A , the band is determined as a conversational sound (voice sound). 
     Further, as shown in  FIG. 4B , when the angle θ exists in the area other than the center area, that is, a right area (or a left area), on the LR plane, the probability of the sound being from the front is low, and the sound of the band is determined as a noise sound. 
     Meanwhile, there is a case in which the sound should be determined as a noise even when the angle θ exists in the center area. When the sample points are broadly dispersed on the LR plane as shown in  FIG. 4C , the straight line LL according to the least-squares method becomes a slope around 45° and the angle θ is included in the center area. 
     The case where the dispersion degree is high as described above is the case where noise sounds arrive widely from many or all directions attributable to surrounding reflected sounds. For example, the case corresponds to a case where sounds, including reflected sounds as in the cabin of an airplane, are heard from all the directions. 
     Here, when the dispersion degree is equal to or higher than a predetermined degree, the sound of the band is determined as a noise even when the angle θ is included in the center area. 
     As a specific example, if the sum of squares of a distance when the straight line LL is obtained using a least squares method is equal to or greater than a specific threshold, it can be determined that the dispersion degree is large. The reason for this is that the sum of squares of the distances from the respective samples SPs to the straight line LL becomes small when the plotted dots of samples are concentrated in the center area, and the sum of squares becomes large in the case shown in  FIG. 4C . 
       FIG. 5  shows an example of the control of each of the sound source directional angle analysis units  54 ,  55 , and  56 . 
     Here, when the sound source directional angle analysis unit  54  performs the above-described analysis on the band BD 1 , the angle θ of the straight line LL is included in the center area. Although the sound source directional angle analysis unit  54  controls the MPF  58  using the control signal SG 1  as described above, in this case, the sound of the band BD 1  is determined as a target sound in this case, so that the band BD 1  which centers on the frequency fc 1  is boosted by the MPF  58  as shown in the drawing. 
     Further, when the sound source directional angle analysis unit  55  performs the above-described analysis on the band BD 2 , the angle θ of the straight line LL is in an area other than the center area. Although the sound source directional angle analysis unit  55  controls the MPF  59  using the control signal SG 2 , the sound of the band BD 2  is determined as a noise in this case, so that the band BD 2  which centers on the frequency fc 2  is attenuated by the MPF  59  as shown in the drawing. 
     Further, when the sound source directional angle analysis unit  56  performs the above-described analysis on the band BD 3 , the angle θ of the straight line LL is included in the center area. However, since the dispersion degree of the sample dots becomes equal to or greater than a predetermined degree, the sound of the band BD 3  is determined as a noise. Although the sound source directional angle analysis unit  56  controls the MPF  60  using the control signal SG 3 , the sound of the band BD 3  is determined as a noise in this case, so that the band BD 3  which centers on the frequency fc 3  is attenuated by the MPF  60  as shown in the drawing. 
     The filter characteristics of the MPFs  58 ,  59 , and  60  are variably controlled based on the above-described determination of the sound source direction for the respective bands, so that the output signal Sout processed by the MPFs  58 ,  59 , and  60  becomes a sound signal in which a sound from the front is boosted and the other noise is attenuated. 
     The processes of the above-described sound source directional angle analysis units  54 ,  55 , and  56  are performed as shown in  FIG. 6 . The process of the sound source directional angle analysis unit  54  will be described. 
     First, the sound source directional angle analysis unit  54  plots the input values of the sound signals SmL and SmR of the band BD 1  to be input on the above-described LR plane during a specific unit time in steps F 101  and F 102 . 
     After plotting a plurality of sample dots during the unit time, the sound source directional angle analysis unit  54  proceeds to step F 103 , obtains a straight line LL using a least-squares method, and then obtains the angle θ of the straight line LL. 
     In step F 104 , it is first determined whether the angle θ is included in the center area or not. If the angle θ is not included in the center area, the sound source directional angle analysis unit  54  proceeds to step F 107 , and then determines that the sound of the corresponding band BD 1  is a noise. Thereafter, the sound source directional angle analysis unit  54  allows the MPF  58  to perform an attenuation process on the band BD 1  using the control signal SG 1 . 
     Meanwhile, it can be considered that the amount of the attenuation in this case becomes, for example, the amount of the attenuation according to the difference between the angle θ at this time and the central angle (for example, 45° of the center area. 
     On the other hand, if it is determined that the angle θ is included in the center area in step F 104 , the sound source directional angle analysis unit  54  proceeds to step F 105  and determines whether the dispersion state is equal to or greater than a specific level. As described above, it may be determined whether the sum of squares of the distance between each of the samples and the straight line LL is equal to or greater than a specific threshold. 
     When the dispersion state is equal to or greater than the specific value, the sound source directional angle analysis unit  54  proceeds to step F 108 , and determines that the sound of the corresponding band BD 1  is a noise. Thereafter, the sound source directional angle analysis unit  54  allows the MPF  58  to perform the attenuation process on the band BD 1  using the control signal SG 1 . 
     Meanwhile, the amount of attenuation in this case can be considered as, for example, the amount of attenuation based on the value of the sum of squares of the distance. 
     When it is determined that the angle θ is included in the center area and the dispersion state is not equal to or greater than the specific value, the sound source directional angle analysis unit  54  proceeds to step F 106 , and determines that the sound of the corresponding band BD 1  is a target sound. Thereafter, the sound source directional angle analysis unit  54  allows the MPF  58  to perform a boosting process on the band BD 1  using the control signal SG 1 . 
     Meanwhile, the amount of boosting in this case can be considered as, for example, the amount of boosting based on the difference between the angle θ at this time and the central angle (for example, 45°) of the center area and based on the dispersion degree. 
     That is, the amount of boosting is greater as the angle θ becomes closer to 45°, and the amount of boosting is greater as the dispersion degree becomes smaller. 
     When any control is performed in steps F 106 , F 107 , and F 108 , the sound source directional angle analysis unit  54  clears the plotted samples in step F 109 , and then returns to step F 101  and performs plotting during the unit time again. Thereafter, the same process is repeated. 
     The sound source directional angle analysis unit  54  repeatedly and continuously performs the above-described process. The sound source directional angle analysis unit  55  and  56  perform the same process. 
     Therefore, for each unit time, the sound source direction of each band is determined and the MPFs  58 ,  59 , and  60  control filter characteristics based on the determination. 
     As understood from the above description, the noise suppression device  1  in the present example divides the input sound signals SmL and SmR into bands BD 1 , BD 2 , and BD 3  using the band pass filters  51 L,  52 L,  53 L,  51 R,  52 R, and  53 R. Thereafter, analysis is performed to determine whether the sound signals are noises or not for the respective bands BD 1 , BD 2 , and BD 3  using the sound source directional angle analysis units  54 ,  55 , and  56 . On the other hand, the sound signals SmL and SmR are added and supplied to the MPFs  58 ,  59 , and  60  which are connected in series. The filter characteristic of each of the MPFs  58 ,  59 , and  60  is variably controlled based on the determination results of the sound source directional angle analysis units  54 ,  55 , and  56 . 
     In this case, the control of a sound stream is a serial filter process and has the same system as a so-called equalizer. Therefore, the deterioration in sound quality due to phase mismatch generated in the above-described configuration of  FIG. 12  does not occur in principle. Therefore, the output signal Sout which has no deterioration in sound quality can be obtained. 
     Further, since an FFT process is not used, the amount of calculation can be suppressed to be low. 
     Further, the band pass filters can be scalably designed according to the amount of resources to be used. It is difficult to be realized in a process which uses FFTs. 
     In addition, the whole system can be mounted with little delay, and can be applied to a field, in particular, to voice communication or the like, which requests an extremely rapid response. 
     2. Example Applied to NC Headphone 
     An example in which the noise suppression device  1  according to the above-described embodiment is applied to a noise cancellation headphone  10  will be described. 
       FIG. 7  schematically shows the noise cancellation headphone (NC headphone)  10  used while being connected to a music reproduction device such as a portable media player  20  or the like. 
     The media player  20  reproduces data, such as music or the like, recorded in an internal recording medium, and outputs sound signals of two L and R channels to the connected NC headphone  10 . 
     The NC headphone  10  includes a headphone unit  11  and a noise cancellation unit  14 . 
     The headphone unit  11  includes the speakers  13 L and  13 R of the L and R channels in respective speaker housings corresponding to both left and right ears of a user. 
     In the case of this example, a noise cancellation process according to a so-called feedforward method is performed, and microphones  2 L and  2 R are provided to collect the external sounds of the respective left and right speaker housings. 
     Meanwhile, the headphone unit  11  may not be the type having the speaker housing as shown in the drawing but may be the types such as an earphone type or an ear pad type. In the case of the present example, the microphones  2 L and  2 R may be provided in any case. 
     The noise cancellation unit  14  is connected to the headphone unit  11  to which the microphones  2 L and  2 R are provided as described above. A monitor switch  43  is provided in the noise cancellation unit  14 , so that a user can perform the on/off operation of a monitor mode. 
     Meanwhile, the monitor mode referred to here is a mode that enables a conversational sound or the like to be successfully heard while the output of music or the like which is being reproduced in the media player  20  is stopped and the noise cancellation function is turned on. 
     The noise cancellation unit  14  mixes a sound signal, such as reproduction music or the like, supplied from the media player  20  with a noise reduction sound signal, so that the sound signal from which the external noises are reduced is output from the speakers  13 L and  13 R. 
     Briefly speaking, noise reduction is performed as follows. 
     The microphones  2 L and  2 R mounted on the speaker housing collect an external noise which reaches the ears of a user through the speaker housing. The noise cancellation unit  14  generates a noise reduction sound signal which has an acoustically reserved phase with respect to that of the external noise based on the sound signal of the external noise collected by the microphones  2 L and  2 R. Thereafter, the noise cancellation unit  14  combines the generated noise reduction sound signal with the sound signal, such as reproduction music or the like, and then supplies the resulting signal to the speakers  13 L and  13 R. 
     Therefore, since the reserved phase component of the external noise is included in the sound output from the speakers  13 L and  13 R, the reserved phase component and the external noise actually leaked through the speaker housing spatially are balanced out, so that the external noise component is reduced and the output sound of the original reproduction music reaches the sense of hearing of the user. 
     The internal configuration of the noise cancellation unit  14  is shown in  FIG. 8 . 
     The noise cancellation unit  14  includes microphone amplifiers  3 L and  3 R, A/D converters  4 L and  4 R, a main processing unit  33  based on a DSP or a CPU, a memory unit  40 , power amplifiers  42 L and  42 R, A/D converters  41 L and  41 R, and a monitor switch  43 . 
     In the main processing unit  33 , a noise cancellation unit  34 , a gain unit  35 , adders  36 L and  36 R, a noise suppression device  1 , a control unit  38 , an equalizer  39 , and switches SW 1  and SW 2  are provided. 
     First, a sound signal, such as reproduction music or the like, from the media player  20  is processed as follows. 
     The reproduction sound signals SA-L and SA-R of the L and R channels are supplied from the media player  20  as a so-called headphone output. 
     The reproduction sound signals SA-L and SA-R are converted into digital signals by the A/D converters  41 L and  41 R. Thereafter, the equalizer  39  performs sound quality correction such as amplitude-frequency characteristic correction, phase-frequency characteristic correction, or both corrections. 
     The correction process of the equalizer  39  is performed based on a control signal from the control unit  38 . For example, the indication of a frequency characteristic or the like is performed using the control signal. 
     The reproduction sound signals SA-L and SA-R in which the sound qualities thereof are corrected by the equalizer  39  are respectively provided to the adders  36 L and  36 R through the switches SW 1  and SW 2  connected to a Te terminal. Thereafter, the reproduction sound signals SA-L and SA-R are added to noise reduction sound signals by the adders  36 L and  36 R, and then the resulting signals are supplied to the power amplifiers  42 L and  42 R. 
     The power amplifiers  42 L and  42 R may be configured with digital amplifiers, and may be configured with a D/A converter and an analog amplifier. 
     Further, the output from the power amplifiers  42 L and  42 R serve as driving signals corresponding to the speakers  13 L and  13 R, and sounds are output from the speakers  13 L and  13 R based on the reproduction sound signals SA-L and SA-R. 
     On the other hand, a process for the above-described noise cancellation is performed as follows. 
     The sound signals SmL and SmR collected by the microphones  2 L and  2 R are amplified by the microphone amplifiers  3 L and  3 R of the noise cancellation unit  14 , and then converted into digital signals by the A/D converters  4 L and  4 R. 
     The sound signals SmL and SmR which are converted into the digital signals and output from the A/D converters  4 L and  4 R are supplied to the noise cancellation unit  34 . The noise cancellation unit  34  serves as a digital filter that generates the above-described noise reduction sound signals using the feedforward method. The noise cancellation unit  34  performs a filtering process on the respective sound signals SmL and SmR using a filter coefficient as instructed by the control signal from the control unit  38 , and then generates the noise reduction sound signals of the L and R channels. 
     The generated noise reduction sound signals of the L and R channels are supplied to the gain unit  35 . The gain unit  35  assigns gains corresponding to the noise reduction sound signals of the L and R channels using the gain coefficient as instructed by the control signal from the control unit  38 . 
     Thereafter, the noise reduction sound signals of the L and R channels from the gain unit  35  are added to the respective reproduction sound signals SA-L and SA-R which are supplied to the adders  36 L and  36 R as described above. 
     The reproduction sounds are output from the speakers  13 L and  13 R based on the reproduction sound signals SA-L and SA-R to which the noise reduction sound signals are added, so that the above-described noise reduction function is exerted. 
     The control unit  38  controls the whole noise cancellation unit. For example, the control unit  38  controls the equalizer  39 , the noise cancellation unit  34 , and the gain unit  35  using the control signal as described above. Further, the control unit  38  can transmit the control signal to the media player  20 . Further, the control unit  38  controls the switching of the switches SW 1  and SW 2 . 
     The memory unit  40  stores information which is referred to when the control unit  38  performs a control process. For example, the memory unit  40  stores information about the filter coefficients of the noise cancellation unit  34  and the equalizer  39  or the like. 
     The noise cancellation unit  14  of the present example further includes the noise suppression device  1  which has the configuration as described in  FIG. 1 . 
     The sound signals SmL and SmR which are converted into digital signals and output from the A/D converters  4 L and  4 R are supplied to the noise suppression device  1 . The noise suppression device  1  performs the configuration and operation described with reference to  FIGS. 1 to 6  on the input sound signals SmL and SmR. 
     Therefore, the output signal Sout, in which the sound from the front direction is boosted as a target sound, such as a conversational sound or the like, and sounds from the other directions are attenuated, is obtained from the noise suppression device  1 . The output signal Sout is supplied to the Tn terminals of the switches SW 1  and SW 2 . 
     Particularly, in the present example, when the control unit  38  detects that a user turns on a monitor mode using the monitor switch  43 , the control unit  38  performs control as follows. 
     If the monitor mode is turned on, the control unit  38  switches the switches SW 1  and SW 2  to the Tn terminals. Meanwhile, when the monitor mode is turned off, the control unit  38  connects the switches SW 1  and SW 2  to the Te terminals, and reproduction music is output from the speakers  13 L and  13 R. 
     Further, the control unit  38  instructs the media player  20  to stop the reproduction music. Therefore, the media player  20  stops the reproduction music. 
     When the control unit  38  performs the above-described control, the output signal Sout of the noise suppression device  1  is supplied to the adder  36 L and  36 R. 
     Therefore, the noise reduction sound signals from the gain unit  35  and the output signal Sout from the noise suppression device  1  are added by the adders  36 L and  36 R, and then the resulting signals are supplied to the power amplifiers  42 L and  42 R. Thereafter, the resulting signals are output from the speakers  13 L and  13 R as sounds. 
     These sounds become speaker output sounds in which, for example, a conversational sound from the front direction is clearly audible while surrounding noises are reduced as in the monitor mode. 
     When the noise suppression device  1  according to the present embodiment is mounted on the NC headphone  10  as described above, the speaker output in which a conversational sound is clearly audible can be realized as a monitor mode operation. 
     That is, when the NC headphone  10  is used, not only noise but also the sound of a person is reduced. However, with the above-described configuration, surrounding noises can be reduced while the sound of a person who is at the front which is equidistance from the microphones  2 L and  2 R is not reduced. Therefore, conversation can be performed more pleasantly while wearing the NC headphone  10 . 
     On that basis, in the noise suppression device  1 , control on a sound stream is performed by a series filtering process as described above, and the deterioration in sound qualities due to phase mismatching does not occur, so that sound can be output without deterioration in sound quality. 
     Further, with a low calculation amount and a process using low resources, it is suitable for mounting on a small device such as the noise cancellation unit  14  or the like. 
     In addition, the whole system can be mounted with low delay. 
     With respect to the monitor mode function of the NC headphone  10 , an actual direct sound and the sound obtained after the process of the noise suppression device  1  is performed are spatially superimposed and then reach the ears of a user. Therefore, if the process delay is large, the sounds are heard as unpleasant echo. However, since the noise suppression device  1  can perform a process with low delay, such an unpleasant echo can be avoided. 
     3. Examples Applied to Various Kinds of Apparatus and Modified Example 
     The noise suppression device  1  according to the present embodiment can be applied to further various kinds of apparatus. 
     For example, it can be considered that the noise suppression device  1  is used for a transmission noise reduction function of a mobile phone. 
     By mounting the noise suppression device  1  on a headset for a mobile phone, the sound can be transmitted to a counter side while the sound, emitted from the mouth of a user which is at equidistance from the microphones, is not reduced and surrounding noises are reduced. 
     Of course, the voice communication performed using a Personal Computer (PC) or a television receiver is the same. 
     Further, an application to a sound recognition front-end may be considered. 
     Nowadays, a sound recognition function-attached “automatic translation” or the like, which is used in a mobile phone or a small-sized Personal Computer (PC), has reached a workable level which can be ordinarily used, and, hereinafter, it can be considered that such a function can be used outside. On the other hand, when a sound is input outdoors, noises which deteriorate the accuracy of sound recognition may be input in many cases. 
     Therefore, for example, if a front-end process is performed using the noise suppression device  1  according to the present embodiment in such a way that microphones are attached, for example, on both ends of portable apparatus, an automatic translation system becomes a system with which the user is satisfied. 
     Further, application as a system for extracting a vocal sound or the like can be considered. 
     Although an application as the microphone input is described in the above-described embodiment, application to line input or a music file can be considered. 
     For example, since a vocal sound, a drum sound, or the like is made to be stereotaxically centered in general music, if the noise suppression device  1  according to the present embodiment is applied thereto, the vocal sound and the drum sound can be separated. Thereafter, of course, if bands are divided, the separation of the vocal sound and the drum sound can be performed. 
     A modified example of the embodiment can be variously considered. 
       FIG. 9  shows a modified example of the configuration of the noise suppression device  1 . This is an example in which a group of two independent systems of series filters is provided in an L channel and an R channel. 
     That is, the sound signal SmL of the L channel is input to series filter systems of MPFs  58 L,  59 L, and  60 L. The sound signal SmR of the R channel is input to series filter systems of MPFs  58 R,  59 R, and  60 R. 
     The filter characteristics of the MPFs  58 L and  58 R are variably controlled using a control signal SG 1  based on the determination of a sound source directional angle analysis unit  54 . 
     The filter characteristics of the MPFs  59 L and  59 R are variably controlled using a control signal SG 2  based on the determination of a sound source directional angle analysis unit  55 . 
     The filter characteristics of the MPFs  60 L and  60 R are variably controlled using a control signal SG 3  based on the determination of a sound source directional angle analysis unit  56 . 
     That is, the modified example is a configuration example in which the operation thereof is the same as in the configuration of  FIG. 1  but output signals SoutL and SoutR of the two L and R channels are output as signals obtained after a process is performed. 
     The noise suppression device  1  may be applied to various types of apparatus with such a configuration. 
     Further, although not shown in the drawing, sound source direction determination to be performed on microphone input sounds of three or more channels can be considered. 
     In such a case, sound signals on which a series filter processes is performed may be combined into a single channel or two channels as shown in  FIG. 8 . Further, the output signals Souts of three or more channels may be obtained in such a way that the series filter process is performed on each of the microphone input sound signals of three or more channels, independently. 
     Further, it may be considered that the sound signal of a single channel is supplied from among the input sound signals of a plurality of channels when a single series filter system is provided. For example, it may be a configuration in which only the sound signal SmL is supplied to the filter group of the MPFs  58 ,  59 , and  60  and the output signal Sout is obtained when the single series filter system (MPFs  58 ,  59 , and  60 ) is provided as shown in  FIG. 1 . 
     Further, it is proper that the number of bands divided by the band pass filters and the bandwidth of a single band are set according to the apparatus to be mounted, a target sound, a usage form, or the like. The number of MPFs connected in series is basically set according to the number of bands divided by the band pass filters. 
     Further, although the process of boosting the sound from the front direction or the back direction as a target sound is described in the embodiment, for example, a process of boosting a sound from the right side as a target sound and reducing sounds from the other directions can be performed. For example, when the angle θ corresponds to the right area as shown in  FIG. 4B , a boosting process may be performed on the MPF corresponding to the band of the target sound, and an attenuating process may be performed on the MPFs corresponding to the bands in which the angle θ corresponds to the center area and the left area. 
     That is, the sound source direction of a target sound can be set using any method. 
     Further, although the noise suppression device  1  performs the digital data process using the A/D converters  4 L and  4 R in the embodiment as shown in  FIG. 1 , the filtering process performed by the MPFs  58 ,  59 , and  60  or the band division performed by the band pass filters may be performed using an analog signal process. 
     The present disclosure contains subject matter related to that disclosed in Japanese Priority Patent Application JP 2010-125502 filed in the Japan patent office on Jun. 1, 2010, the entire contents of which are hereby incorporated by reference. 
     It should be understood by those skilled in the art that various modifications, combinations, sub-combinations and alterations may occur depending on design requirements and other factors insofar as they are within the scope of the appended claims or the equivalents thereof.