Patent Publication Number: US-9894455-B2

Title: Correction of sound signal based on shift of listening point

Description:
BACKGROUND 
     The present technology relates to a sound processing device and a method thereof, a program, and a recording medium, and particularly to a sound processing device and method, a program, and a recording medium that enable simple field correction at an arbitrary position of a sound field space. 
     In a multi-channel audio system, from peripheral speakers disposed in different positions, sounds of different channels corresponding to the positions are emitted, and thus, a listener can enjoy music enriched with the same sense of presence as if the listener is in a theater or at a music hall. 
     In general, an ideal listening position is the center position in the disposition of speakers, and if a listening position is shifted from the center, the listener is not able to enjoy music in high sound quality. 
     Thus, correcting a sound field according to the position of a listener has been proposed (for example, refer to Japanese Unexamined Patent Application Publication No. 2006-287606). In the technology disclosed in Japanese Unexamined Patent Application Publication No. 2006-287606, an infrared light emitting unit is installed in a microphone, and infrared light sensing units are installed in respective speakers. An infrared light beam emitted from the infrared light emitting unit of the microphone disposed in a position of a listener is sensed by the infrared light sensing units installed in the speakers. The distance between each speaker and the microphone is computed from a level of sensing the infrared light, and the position of the microphone, i.e., the position of the listener is computed based on the distance. Further, a sound field is corrected based on the position. 
     SUMMARY 
     However, in the case of the technology disclosed in Japanese Unexamined Patent Application Publication No. 2006-287606, when a listening position is changed, it is necessary to accordingly measure the characteristics of a sound field thoroughly again, and the correction of the sound field takes time, which are inconvenient for a user. 
     It is desirable to enable simple correction of a sound field at an arbitrary position in a sound field space. According to an embodiment of the present technology, there is provided a sound processing device which includes a shift detection unit that detects a shift of a listening point on which a user listens to a sound in a sound field space from a standard reference listening point, and a correction unit that corrects a sound signal based on first sound field correction data for correcting a sound field when the listening point is the standard reference listening point and second sound field correction data for correcting a sound field of the listening point when the listening point is shifted from the standard reference listening point. 
     According to the embodiment, the second sound field correction data may be data of a peripheral reference listening point in a position shifted from the standard reference listening point. 
     According to the embodiment, when the position of the listening point is different from the position of the peripheral reference listening point, the second sound field correction data on the nearest peripheral reference listening point may be the second sound field correction data of the listening point. 
     According to the embodiment, the shift of the listening point from the standard reference listening point may be detected by collecting a measurement sound generated from the listening point. 
     According to the embodiment, the shift of the listening point from the standard reference listening point may be detected from the position of the listening point with respect to a position in which the measurement sound generated from the listening point is collected and a position in which the measurement sound is collected with respect to the standard reference listening point. 
     According to the embodiment, the position of the listening point with respect to the position in which the measurement sound generated from the listening point is collected may be detected from a sound signal obtained by collecting the measurement sound generated from the listening point. 
     According to the embodiment, the position in which the measurement sound is collected with respect to the standard reference listening point may be detected from position information of a sound emitting unit that emits a sound that the user listens to on the listening point in the sound field space with respect to the position in which the measurement sound is collected and position information of the sound emitting unit with respect to the standard reference listening point. 
     According to the embodiment, a measurement unit that obtains the first sound field correction data by measuring a sound field of the sound field space may be further included. 
     According to the embodiment, a storage unit that stores the second sound field correction data in advance may be further included. 
     According to the embodiment, the storage unit may further store the first sound field correction data measured by the measurement unit. 
     According to the embodiment, a parameter calculated from the first sound field correction data and the second sound field correction data may be set in the correction unit. 
     According to the embodiment, the first sound field correction data and the second sound field correction data may be set as individual parameters in different correction units. 
     According to another embodiment of the present technology, there is provided a sound processing method that includes steps of detecting a shift of a listening point on which a user listens to a sound in a sound field space from a standard reference listening point, and correcting a sound signal based on first sound field correction data for correcting a sound field when the listening point is the standard reference listening point and second sound field correction data for correcting a sound field of the listening point when the listening point is shifted from the standard reference listening point. 
     According to still another embodiment of the present technology, there is provided a program that causes a computer to execute processes including steps of detecting a shift of a listening point on which a user listens to a sound in a sound field space from a standard reference listening point, and correcting a sound signal based on first sound field correction data for correcting a sound field when the listening point is the standard reference listening point and second sound field correction data for correcting a sound field of the listening point when the listening point is shifted from the standard reference listening point. 
     According to still another embodiment of the present technology, there is provided a recording medium on which is recorded a program that causes a computer to execute processes including steps of detecting a shift of a listening point on which a user listens to a sound in a sound field space from a standard reference listening point, and correcting a sound signal based on first sound field correction data for correcting a sound field when the listening point is the standard reference listening point and second sound field correction data for correcting a sound field of the listening point when the listening point is shifted from the standard reference listening point. 
     According to the embodiments of the present technology, a shift of a listening point on which a user listens to a sound in a sound field space from a standard reference listening point is detected and a sound signal is corrected based on first sound field correction data for correcting a sound field when the listening point is the standard reference listening point and second sound field correction data for correcting a sound field of the listening point when the listening point is shifted from the standard reference listening point. 
     As described above, according to the embodiments of the present technology, a sound field can be simply corrected at an arbitrary position in a sound field space. 
    
    
     
       BRIEF DESCRIPTION OF THE DRAWINGS 
         FIG. 1  is a block diagram illustrating a configuration of an embodiment of a sound processing system according to the present technology; 
         FIG. 2  is a block diagram illustrating a functional configuration of a CPU that performs a sound field correction process; 
         FIG. 3  is a block diagram illustrating a functional configuration of a DSP that performs a sound field correction process; 
         FIG. 4  is a diagram describing a sound field correction data map; 
         FIG. 5  is a flowchart describing a process of creating a sound field correction data map; 
         FIG. 6  is a diagram illustrating an example of a frequency characteristic of equalizer devices; 
         FIGS. 7A to 7C  are graphs describing examples of an averaging process; 
         FIG. 8  is a flowchart describing an operation of a sound field correction process; 
         FIG. 9  is a diagram describing positions in a sound field space; 
         FIG. 10  is a block diagram illustrating a functional configuration of a CPU that performs a position detection process; 
         FIG. 11  is a flowchart describing the position detection process; 
         FIG. 12  is a diagram illustrating an example of a result of sound source direction estimation; 
         FIG. 13  is a diagram illustrating an example of disposition of microphones; 
         FIG. 14  is a diagram illustrating an example of a function of a linear equation of a line passing through a center point between a sound source and microphones; 
         FIG. 15  is a diagram illustrating an example of a function of the linear equation of a line passing through the center point between the sound source and microphones; 
         FIG. 16  is a diagram illustrating an example of a function of the linear equation of a line passing through the center point between the sound source and microphones; 
         FIG. 17  is a flowchart showing a position detection process on a two-dimensional plane; and 
         FIG. 18  is a block diagram illustrating another configuration example of a correction unit. 
     
    
    
     DETAILED DESCRIPTION OF EMBODIMENTS 
     Hereinafter, embodiments for implementing the present technology (hereinafter, referred to as embodiments) will be described. Note that description will be provided in the following order: 
     1. Configuration of Sound Processing System  1   
     2. Functional configuration of CPU  21   
     3. Functional configuration of DSP  22   
     4. Sound field correction data map 
     5. Sound field correction process 
     6. Position detection process 
     7. Modification example 
     8. Application of the present technology to a program 
     9. Other configurations 
     Configuration of Sound Processing System  1   
       FIG. 1  is a block diagram illustrating a configuration of a sound processing system  1  according to an embodiment of the present technology. This sound processing system  1  includes a reproduction device  11 , a sound processing device  12 , and a sound field space  13  as shown in  FIG. 1 . 
     The reproduction device  11  reproduces recorded information from a recording medium, for example, an optical disc such as a Blu-ray Disc (BD) (registered trademark), a High Definition Digital Versatile Disc (HD-DVD), a hard disk, or a semiconductor memory. In addition, the reproduction device  11  reproduces received information using a built-in tuner. The sound field space  13  is a space in which a user listens to a sound such as music as a reproduced sound. In the present embodiment, there are three microphones  31   a ,  31   b , and  31   c , and eight speakers  32   1  to  32   8  disposed in the sound field space  13 . Note that, when it is not necessary to separately distinguish the microphones or the speakers hereinbelow, they are simply referred to as the microphones  31  or the speakers  32 . 
     The sound processing device  12  includes a CPU (Central Processing Unit)  21 , a DSP (Digital Signal Processor)  22 , a DIR (Digital Interface Receiver)  23 , an amplifier  24 , an A/D (Analog to Digital) converter  25 , an operation unit  26 , an amplifier  27 , a non-volatile memory  28 , and a removable medium  29 . 
     The speakers  32  are disposed in predetermined positions in the sound field space  13 . In the case of a 7.1 channel speaker system, for example, a front-left speaker  32   2  is disposed on the left side, a front-right speaker  32   3  is disposed on the right side, and a center speaker  32   1  is disposed at the center in front of a standard reference listening point P 0 . In addition, a surround-left speaker  32   4  is disposed on the left side and a surround-right speaker  32   5  is disposed on the right side on the back side of the listening position, and a surround-back left speaker  32   6  is disposed on the left side and a surround-back right speaker  32   7  is disposed on the right side between them. Further, a sub-woofer speaker  32   8  dedicated to reproduction of a low frequency sound is disposed in an arbitrary position (in the present embodiment, on the left side of the center on the front side). 
     In the sound processing system  1 , the user measures a sound field characteristic of the sound field space  13  in advance when the user listens to music, or the like in the sound field space  13 . This measurement is performed basically once, but when a listening environment is changed due to a change in the disposition of the speakers  32 , the measurement is performed according to the change. 
     When the sound field characteristic of the sound field space  13  is measured, a measurement signal for measuring the sound field characteristic is supplied to the plurality of speakers  32  disposed in the predetermined positions in the sound field space  13 . This measurement signal can be set to be, for example, an impulse signal. Then, a sound output from the speakers  32  is collected by the plurality of microphones  31 . Then, based on measured data computed from the sound collecting signal obtained as above, an acoustic characteristic and a frequency characteristic of the plurality of speakers  32  are corrected, and a virtual sound image localization process is performed. In other words, a sound field correction process is performed. 
     When the sound field characteristic of the sound field space  13  is measured, the microphones  31  are disposed in the standard reference listening point P 0  being separate from each other by a predetermined distance. The standard reference listening point P 0  is a standard point among arbitrary listening points at which a user actually listens to a heard sound such as music. To be specific, the three microphones  31  are disposed in, for example, an L shape. In other words, the microphones are disposed so that the line connecting the microphone  31   a  and the microphone  31   b  and the line connecting the microphone  31   a  and the microphone  31   c  form substantially a right angle, in other words, the three microphones  31   a  to  31   c  are positioned at the vertexes of a right-angle isosceles triangle. In addition, the three microphones  31   a  to  31   c  are disposed so that the standard reference listening point P 0  is positioned at the center of the bottom side of the right-angle isosceles triangle. Of course, the disposition of the microphones  31  is not limited thereto, and any disposition is possible as long as the standard reference listening point P 0  can be specified from positions in which the microphones  31  are disposed. 
     The operation unit  26  is a user interface (UI) with which a user operates the CPU  21 , and is input with instructions of, for example, a sound field characteristic measurement mode, a position detection mode, a reproduction mode, and the like. 
     The CPU  21  controls each unit having a random access memory (RAM), which is not shown, as a work memory according to a program stored in a read-only memory (ROM) or the removable medium  29 , which is not shown, in advance. When the position detection mode is instructed from the operation unit  26 , for example, the CPU  21  controls the DSP  22  to output a measurement signal for detecting a position and to output a measurement signal from the amplifier  27 . In addition, the CPU  21  causes the non-volatile memory  28  to store measurement data computed from the DSP  22  based on a measurement sound collected by the microphones  31 , and controls the DSP  22  based on the above measurement data during reproduction of a sound signal of a heard sound. The amplifier  27  amplifies a measurement signal supplied from the DSP  22  and a sound signal of a heard sound processed in the DSP  22 , and supplies the signals to the speakers  32 . 
     The microphones  31  collect measurement sound output from the speakers  32  to which the measurement signal is supplied, convert the collected measurement sound into a sound collecting signal, and supply the signal to the amplifier  24 . The amplifier  24  amplifies the sound collecting signal from each of the microphones  31 , and supplies the result to the A/D converter  25 . The A/D converter  25  converts the sound collecting signal from each of the microphones  31  that has been amplified in the amplifier  24  into a digital signal, and supplies the signal to the DSP  22 . 
     When a sound signal from the reproduction device  11  is reproduced, the CPU  21  performs a sound field correction process. In other words, the non-volatile memory  28  stores sound field correction data, and the CPU  21  reads the sound field correction data from the non-volatile memory  28  and sets the data in the DSP  22 . Accordingly, the sound field correction process is performed on the sound signal supplied from the reproduction device  11  via the DIR  23 , and the processed sound signal is amplified by the amplifier  27 , and output from the speakers  32 . 
     Note that the non-volatile memory  28  stores two kinds of sound field correction data. First sound field correction data is sound field correction data obtained as a result of measuring the sound field space  13  in the sound field characteristic measurement mode. Second sound field correction data is data for position adjustment. When an actual listening point of a user is changed to another point from the standard reference listening point P 0 , the sound field correction data for position adjustment is data for enabling the same sound field characteristic as in the standard reference listening point P 0  to be realized. The sound field correction data for position adjustment is prepared in advance by the manufacturer of the sound processing device  12 , and stored in the non-volatile memory  28  as a sound field correction data map  101  (to be described later with reference to  FIG. 4 ). 
     Functional Configuration of the CPU  21   
       FIG. 2  is a block diagram illustrating a functional configuration of the CPU  21  that performs a sound field correction process. As illustrated in the drawing, the CPU  21  has functional blocks of a setting unit  41 , a determination unit  42 , a detection unit  43 , a specification unit  44 , an acquisition unit  45 , and a measurement unit  46 . 
     The setting unit  41  sets the sound field correction data. The determination unit  42  performs a determination process. The detection unit  43  detects a position. The specification unit  44  performs a process of specifying a position. The acquisition unit  45  acquires data. The measurement unit  46  executes a sound field characteristic measurement process of the sound field space  13 . 
     Functional Configuration of the DSP  22   
       FIG. 3  is a block diagram illustrating a functional configuration of the DSP  22  that performs a sound field correction process. The DSP  22  has a correction unit  71 . The correction unit  71  includes a sound field processing device  61 , equalizer devices  62   1  to  62   8 , and delay devices  63   1  to  63   8 . When it is not necessary to distinguish the individual equalizer devices  62   1  to  62   8  and delay devices  63   1  to  63   8 , they are referred to simply as the equalizer devices  62  and the delay devices  63 . 
     The sound field processing device  61  performs sound field processing such as reverberation addition onto a sound signal of each channel supplied from the DIR  23 , and supplies the result to the equalizer devices  62  of corresponding channels. The sound field processing is a process of generating an invisible virtual speaker  121  (to be described later with reference to  FIG. 9 ), and reproducing a space reverberation characteristic of a predetermined studio or a theater, or the like. The equalizer devices  62  are so-called parametric equalizers, correcting a frequency characteristic of a sound signal of each channel supplied from the sound field processing device  61  based on a center frequency f 0 , a gain G, and an acutance Q which are parameters given from the CPU  21 . The corrected sound signal is supplied to the delay devices  63  of corresponding channels. 
     Note that, as in the embodiment of  FIG. 3 , providing one correction unit  71  can simplify the configuration more than providing two correction units  71  as in the embodiment of  FIG. 18  to be described later. 
     Sound Field Correction Data Map 
     A sound field correction data map provided by a manufacturer is stored in advance in the non-volatile memory  28 . 
       FIG. 4  is a diagram describing a sound field correction data map. Note that reference numerals of units in the sound field correction data map  101  are indicated with the addition of an apostrophe (′) to corresponding reference numerals of units in the sound field space  13 . 
     As illustrated in  FIG. 4 , the sound field correction data map  101  is a map of sound field correction data of a reference listening point P′ j  in a listening space under an ideal environment in which a plurality of speakers  32 ′ are disposed. The reference listening points P′ j  (j=0, 1, 2, . . . , n) are composed of a standard reference listening point P′ 0  and peripheral reference listening points P′ i  (i=1, 2, . . . , n). The standard reference listening point P′ 0  is a point of a standard position for listening to music, or the like in the listening space under the ideal environment. 
     Based on the standard reference listening point P′ 0  serving as the center, a plurality of lines  111  in a concentric circular shape disposed at predetermined intervals and a plurality of lines  112  in a radial shape are imagined. In addition, reference listening points P′ j  positioned on the intersection of the plurality of lines  111  and lines  112  are peripheral reference listening points P′ i . In other words, the plurality of peripheral reference listening points P′ i  are present in the peripheries shifted from the standard reference listening point P′ 0 . 
     The sound field correction data (in other words, the second sound field correction data) of the sound field correction data map  101  is data necessary for forming the same sound field as the standard reference listening point P′ 0  when a listening point is shifted from the standard reference listening point P′ 0 . For example, it is assumed that a parameter necessary for generating a sound field that can realize a sufficient sense of presence at the standard reference listening point P′ 0  intended in designing (hereinafter, referred to as a target sound field) is PR′ 0 . In other words, if the parameter PR′ 0  is set in the correction unit  71 , the virtual speaker  121  (to be described later with reference to  FIG. 9 ) is generated in a predetermined position, and a sound is output from the speaker, realizing the target sound field. In the same manner, it is assumed that a parameter necessary for generating a target sound field at a peripheral reference listening point P′ i  is PR′ i . In this case, sound field correction data (in other words, second sound field correction data) D′ i  of the peripheral reference listening point P′ i  on the sound field correction data map  101  is computed from the following formula:
 
 D′   i   =PR′   i   −PR′   0   (1)
 
     In other words, the sound field correction data D′ i  is a parameter of the difference between the parameter PR′ i  and the parameter PR′ 0 . If the sound field correction data D′ i  of the formula (1) is expressed when including a case of the standard reference listening point P′ 0 , the sound field correction data D′ 0  on the sound field correction data map  101  at the standard reference listening point P′ 0  becomes 0 as in the formula (2):
 
 D′   j   =PR′   j   −PR′   0   (2)
 
     Next, with reference to  FIG. 5 , a process of creating a sound field correction data map performed by a manufacturer will be described exemplifying sound field correction on a phase. Herein, a case in which sound field correction data of sound signals of front-right and -left channels at the standard reference listening point P′ 0  will be described for the sake of simplicity. Note that, although not shown in the drawing, the manufacturer has a sound processing system  1 ′ the same as the sound processing system  1  illustrated in  FIG. 1  in order to create the sound field correction data map. Hereinbelow, reference numerals obtained by adding apostrophes (′) to the reference numerals of the units of the sound processing system  1  of  FIG. 1  are used as corresponding reference numerals of units of the sound processing system  1 ′. 
       FIG. 5  is a flowchart describing the process of creating a sound field correction data map. In Step S 31 , a process of measuring a sound field is performed. In other words, a sound emitted from a pair of right and left speakers  32 ′ disposed in predetermined positions is collected by a microphone  31 ′ disposed on the standard reference listening point P′ 0 . Using the sound collecting signal, measurement data such as a frequency characteristic, and position information of the pair of right and left speakers  32 ′ is obtained for right and left channels. 
     In Step S 32 , an averaging process is performed. In other words, parameters for frequency characteristic correction of the right and left channels included in the measurement data are rearranged in order from lower center frequencies for the right and left channels. Then, average values of the parameters for pairs of the right and left channels are computed in order from lower center frequencies. 
     The averaging process will be described below exemplifying a frequency characteristic of equalizer devices  62 ′ that perform sound field correction. 
       FIG. 6  illustrates an example of the frequency characteristic of the equalizer devices  62 ′. Note that, in  FIG. 6 , the horizontal axis that is a frequency axis is a logarithmic axis. The frequency characteristic is set with parameters of a center frequency f 0 , a gain G, and an acutance Q. The equalizer devices  62 ′ corresponding to channels are configured to correct the frequency characteristic for each band within the corresponding channel. When an audio frequency band of each channel is divided into six bands, for example, parameters are set in each of the equalizer devices  62 ′ for the six bands, and the frequency characteristic for sound signals is corrected. 
     Delay devices  63 ′ cause delay times of the sound signals of each channel supplied from the equalizer devices  62 ′ to change, and correct, for example, the distances between the speakers  32 ′ and the standard reference listening point P′ 0 . 
       FIGS. 7A to 7C  are graphs describing examples of the averaging process. Note that, in  FIGS. 7A to 7C , the horizontal axis that is a frequency axis is indicated by a logarithmic axis. 
     As illustrated in  FIGS. 7A and 7B , parameters are computed for front-right and -left channels. In other words, center frequencies f L1 , f L2 , f L3 , . . . , center frequencies f R1 , f R2 , f R3 , . . . , Gains G L1 , G L2 , G L3 , . . . , Gains G R1 , G R2 , G R3 , . . . , acutances Q L1 , Q L2 , Q L3 , . . . , and acutances Q R1 , Q R2 , Q R3 , . . . are computed. 
     The computed parameters are rearranged in order from lower center frequencies for each front-right and -left channel. Then, as illustrated in  FIG. 7C , average values of the parameters are computed for pairs of the right and left channels in order from lower center frequencies. 
     
       
         
           
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     After the averaging process is performed as described above, in Step S 33 , a process of storing the parameters in a non-volatile memory  28 ′ is performed. In other words, the parameters of the equalizer devices  62 ′ and the delay devices  63 ′ of the front-left channel, and the parameters of the equalizer devices  62 ′ and the delay devices  63 ′ of the front-right channel that have undergone the averaging process in Step S 32  are stored in the non-volatile memory  28 ′. The parameters correct frequency characteristics of the sound signals of the front-right and -left channels to be the same. 
     The same process is executed on other all speakers  32 ′. Accordingly, sound field correction data as a parameter PR 0  necessary for realizing the target sound field on the standard reference listening point P′ 0  is acquired, and associated with position information. 
     After a sound field characteristic measurement process is performed on the standard reference listening point P′ 0 , the microphone  31 ′ is moved to another peripheral reference listening point P′ i , and a sound field characteristic measurement process is executed there. Then, a parameter PR i  necessary for realizing the target sound field on the peripheral reference listening point P′ i  is computed. Further, based on the above formula (2), sound field correction data D′ j  as a parameter of the difference between a parameter PR′ j  and a parameter PR′ 0  is computed. 
     In this manner, the sound field correction data on all of the reference listening points P′ j  is acquired, and stored in the non-volatile memory  28 ′ as the sound field correction data map  101  with the position information. Note that the position information of the speakers  32 ′ is also stored in the non-volatile memory  28 ′ as a part of the sound field correction data map  101 . Coordinates of the speakers  32 ′ are coordinates with reference to the standard reference listening point P′ 0 . 
     Hereinabove, an example of sound field correction with regard to the phase of the speakers  32 ′ has been described, however, sound field correction data such as an equalizer, a gain, or delay in addition to phase is also acquired. In other words, entire sound field correction data necessary for realizing a target sound field is acquired on each peripheral reference listening point P′ i  in the same manner as on the standard reference listening point P′ 0 , and is stored in the non-volatile memory  28 ′ as the sound field correction data map  101 . The sound field correction data map  101  stored in the non-volatile memory  28 ′ is stored in the non-volatile memory  28 . Alternatively, the non-volatile memory  28 ′ is used as the non-volatile memory  28 . 
     Sound Field Correction Process 
     Next, with reference to the flowchart of  FIG. 8 , the sound field correction process by the sound processing system  1  will be described.  FIG. 8  is a flowchart describing an operation of the sound field correction process. 
     In Step S 41 , the sound field characteristic measurement mode is set, and the measurement unit  46  executes a sound field characteristic measurement process in the sound field space  13 . The sound field characteristic measurement process is performed only on the standard reference listening point P 0  that is a standard point on which a user listens to music, or the like in the sound field space  13 . In other words, a microphone  31  is disposed on the standard reference listening point P 0 , a measurement signal for sound field characteristic measurement is supplied to the speakers  32 , and sound field correction data necessary for realizing an ideal sound field on the standard reference listening point P 0  is acquired. The sound field correction data is stored in the non-volatile memory  28 . The sound field characteristic measurement process on the standard reference listening point P 0  in the sound field space  13  is the same as the sound field characteristic measurement process on the standard reference listening point P′ 0  in the process of creating the sound field correction data map, and repetitive detailed description will be omitted. 
     The sound field characteristic measurement process in the sound field space  13  is performed only once for the first time after the plurality of speakers  32  are disposed in predetermined positions in the sound field space  13 , however when the environment is changed such as the positions of the speakers  32  being changed, the process is performed again. 
     The determination unit  42  determines whether or not the microphone  31  is positioned on the standard reference listening point P 0  in the sound field space  13  in Step S 42 . In other words, the user disposes the microphone  31  in arbitrary positions in the sound field space  13 . The microphone  31  can be disposed in positions not interfering with the user listening to music, or the like. Then, the user inputs that the position in which the microphone  31  is disposed is the standard reference listening point P 0  with an operation of the operation unit  26 . The determination unit  42  performs the determination process of Step S 42  based on the input from the user. 
     When the microphone  31  is determined not to be positioned on the standard reference listening point P 0  in Step S 42 , in other words, when the microphones  31  are positioned in positions other than the standard reference listening point P 0 , a position detection mode is set in Step S 43 , and a position detection process is executed. Though details of the position detection process will be described later with reference to  FIG. 11 , the position of the microphone  31  with respect to the standard reference listening point P 0  which is disposed at that time is detected from the process. 
       FIG. 9  is a diagram describing positions in the sound field space  13 . A space  101 A in the sound field space  13  can be considered to correspond to the sound field correction data map  101 . The microphone  31  in  FIG. 9  is disposed in a position (in the vicinity of a line  111 A on the outermost side of the drawing) different from the position of the standard reference listening point P 0 . From the position detection process of Step S 43 , position difference data V 1  denoting a vector facing the microphone  31  from the standard reference listening point P 0  in  FIG. 9  is obtained. The position difference data V 1  is position information indicating the position of the microphone  31  with respect to the standard reference listening point P 0 . 
     On the other hand, when the microphone  31  is determined to be positioned on the standard reference listening point P 0  in Step S 42 , the setting unit  41  sets the position difference data V 1  to be 0 in Step S 44 . 
     After the processes of Step S 43  and Step S 44 , the acquisition unit  45  acquires a signal of a measurement sound from a listening point P k  in Step S 45 . In other words, the user moves to the listening point P k  that is the position in which the user wants to listen to music, or the like. Then, the user generates measurement sounds by, for example, clapping hands or raising a voice. The generated measurement sounds are collected by the microphone  31 . Sound signals of the collected measurement sounds are amplified by the amplifier  24 , A/D-converted by the A/D converter  25 , and acquired by the acquisition unit  45  of the CPU  21  via the DSP  22 . 
     The detection unit  43  detects the position of the listening point P k  with respect to the microphone  31  in Step S 46 . In other words, with the sound signals collected in the process of Step S 45 , the position of the listening point P k  from the microphone  31  that collects the measurement sounds can be detected. This process is the same as the position detection process (process of  FIG. 11  to be described later) of Step S 43 . In other words, the detection unit  43  corresponds to a sound collecting signal acquisition unit  81 , a phase difference calculation unit  82 , a stencil filtering process unit  83 , a sound source direction estimation unit  84 , and a position detection unit  85  of  FIG. 10  to be described later. In this manner, position data V 2  that is a vector toward the listening position P k  from the microphone  31  of  FIG. 9  is obtained. The position data V 2  is position information indicating the position of the listening position P k  with respect to the microphone  31 . 
     The specification unit  44  specifies the position of the listening position P k  with respect to the standard reference listening point P 0  in Step S 47 . In other words, as shown in  FIG. 9 , the position difference data V 1  obtained in Step S 43  or Step S 44  and the position data V 2  obtained in Step S 46  are combined. Accordingly, position data V 3  that is a vector as position information indicating the position of listening position P k  with respect to the standard reference listening point P 0  is obtained. 
     The acquisition unit  45  acquires sound field correction data of the listening position P k  on the sound field correction data map  101  in Step S 48 . As described above, the non-volatile memory  28  stores the sound field correction data map  101 , and on the sound field correction data map  101 , the sound field correction data of the standard reference listening point P 0  and the peripheral reference listening point P i  is associated with the position data. The acquisition unit  45  acquires sound field correction data of the reference listening point P j  corresponding to the listening position P k  specified in the process of Step S 47  from the sound field correction data map  101 . When the position of the listening position P k  specified in the process of Step S 47  is different from the position of the standard reference listening point P 0  or the reference listening point P j , sound field correction data of the nearest reference listening point P j  is acquired as the sound field correction data of the listening position P k . 
     Accordingly, data for performing sound field correction can be simply and quickly obtained not only on the standard reference listening point P 0  in the sound field space  13  but also on an arbitrary listening position P k . In addition, it is not necessary to dispose the microphone  31  on the listening position P k , the microphone  31  can be disposed in an arbitrary position in the sound field space  13 , and work on sound field correction becomes simple, without necessitating rearrangement of the microphone  31 . 
     In Step S 49 , the setting unit  41  sets the sound field correction data acquired in Step S 48 . In other words, the sound field correction data is set as a parameter of the correction unit  71  of the DSP  22  (to be specific, the sound field processing device  61 , the equalizer devices  62 , and the delay devices  63 ) that processes the sound signals of each channel. In this case, the setting unit  41  calculates the parameter as the sound field correction data to be set based on the following formula (3). The formula (3) is a formula elicited when the above-described formula (2) is applied to the sound field space  13  in an ideal environment.
 
 PR   j   =PR   0   +D′   j   (3)
 
     In other words, the condition of a sound field made from a shift between the standard reference listening point P′ 0  and the reference listening points P′ j  in an ideal environment is considered to be the same as (or similar to) the condition of a sound field made from a shift between the standard reference listening point P 0  and the reference listening points P j  in the sound field space  13 . Thus, instead of the parameter PR′ 0  of the standard reference listening point P′ 0  in the ideal environment of the formula (2), the parameter PR 0  of the standard reference listening point P 0  in the sound field space  13  is applied. Then, by modifying the formula (2), a formula for obtaining the parameter PR j  of the reference listening points P j  in the sound field space  13  corresponding to the parameter PR′ j  of the reference listening points P′ j  in the ideal environment is obtained. Thus, the parameter PR 0  (in other words, first sound field correction data) stored in the non-volatile memory  28  in the process of Step S 41  is read, and sound field correction data D′ j  that is second sound field correction data is read from the non-volatile memory  28  in the process of Step S 48 . Then, by applying the data to the formula (3), the parameter PR j  on the listening point P k  can be obtained. 
     The DSP  22  corrects the sound field in Step S 50 . In other words, the sound field correction data PR j  calculated based on the formula (3) as described above is set in the correction unit  71  (the sound field processing device  61 , the equalizer devices  62 , and the delay devices  63 ) as a parameter. Accordingly, a target sound field that is substantially the same as (in other words, similar to) an ideal sound field space provided by the manufacturer is formed on the listening point P k  in the sound field space  13 . As a result, after the formation, a reproduction mode is set, reproduction is instructed to the reproduction device  11 , and the sound signals reproduced from the device and input via the DIR  23  are appropriately corrected by the correction unit  71  of the DSP  22 , and then output from the speakers  32  via the amplifier  27 . Thus, the user can listen to high-quality sound information on the reference listening point P j  in the sound field space  13  and a near arbitrary listening point P k  as well. 
     Position Detection Process 
     Next, details of the position detection process of Step S 43  of  FIG. 8  will be described. In order to perform this process, the CPU  21  further includes the configuration of  FIG. 10 . 
       FIG. 10  is a block diagram illustrating a functional configuration of the CPU  21  that performs the position detection process. The CPU  21  acquires sound collecting signals from the microphones  31  disposed in the predetermined sound field space  13 , and specifies the positions of a sound source and a virtual sound source. Three microphones  31  are provided in this embodiment, but can be provided in an arbitrary number n (n is a natural number equal to or higher than 3). 
     The CPU  21  in the present embodiment includes the sound collecting signal acquisition unit  81 , the phase difference calculation unit  82 , the stencil filtering process unit  83 , the sound source direction estimation unit  84 , and the position detection unit  85 . 
     The sound collecting signal acquisition unit  81  has a function of acquiring a plurality of pairs of sound collecting signals in which sound collecting signals from two microphones  31  among the three microphones  31  are made to be one pair. For example, sound collecting signals of a pair of the first microphone  31   a  and the second microphone  31   b , a pair of the second microphone  31   b  and the third microphone  31   c , and a pair of the first microphone  31   a  and the third microphone  31   c  are acquired. 
     Note that, as will be described later, the accuracy of the microphones  31  improves as the number of the microphones increases, and it is preferable to constitute the microphones in a higher number than the dimension number of the space to be measured. Three microphones are appropriate for specifying the positions of the sound source and the virtual sound source in a two-dimensional space, and four microphones are appropriate for specifying the positions of the sound source and the virtual sound source in a three-dimensional space. 
     In addition, in specification of the positions of a sound source and a virtual sound source in the two-dimensional space, it is preferable that the three microphones  31  be disposed so as to be positioned on the vertexes of a right-angle triangle. Furthermore, in specification of the positions of a sound source and a virtual sound source in a three-dimensional space, it is preferable that four microphones  31  be disposed so as to be positioned on the vertexes of a cube indicating three-dimensional directions. With the configuration as described above, a sound source or a virtual sound source in an arbitrary direction in each dimensional space can be detected striking a balance, and calculation methods of respective directions can be configured to be the same, and thus, a burden of calculation can be reduced, and calculation accuracy can be stably enhanced. 
     Note that, when the positions of a sound source and a virtual sound source in a two-dimensional space are specified, if one of two positions obtained from calculation is selected by a user, the number of microphones  31  can be two. 
     The phase difference calculation unit  82  computes the phase difference of each frequency of the plurality of pairs of acquired sound collecting signals. 
     The stencil filtering process unit  83  has a function of accumulating phase differences periodically occurring for each frequency of the phase difference of the plurality of pairs of sound collecting signals. 
     The sound source direction estimation unit  84  has a function of estimating a plurality of pairs of sound source directions expressed by a linear equation of a line passing through a center point between the microphones  31  in each pair of microphones  31  from the information of the phase difference of each frequency of the sound collecting signals that has undergone the stencil filtering process. As will be described later, the linear equation expressed and estimated from the phase difference of each pair of sound collecting signals is a linear equation of a line passing through a sound source position estimated from the phase difference of each pair of sound collecting signals. 
     The position detection unit  85  has a function of specifying least-squares solutions of simultaneous equations that include the plurality of linear equations as the positions of a sound source and a virtual sound source. The specified positions of the sound source and the virtual sound source can be displayed on a display device (not shown) as two-dimensional or three-dimensional space coordinates if necessary. 
     Next, details of the position detection process executed in Step S 43  of  FIG. 8  will be described with reference to  FIG. 11 .  FIG. 11  is a flowchart describing the position detection process. 
     The sound collecting signal acquisition unit  81  of the CPU  21  acquires sound collecting signals in Step S 61 . In other words, the sound collecting signal acquisition unit  81  controls the DSP  22  to output a reference sound from one selected virtual speaker  121 . The sound collecting signal acquisition unit  81  acquires the sound collecting signals collected by the three microphones  31 . The plurality of pairs of acquired sound collecting signals are converted into digital signals, and stored in a memory, which is not shown. 
     The phase difference calculation unit  82  detects the phase difference of the plurality of pairs of sound collecting signals in Step S 62 . In other words, the phase difference calculation unit  82  reads the plurality of pairs of sound collecting signals stored in the memory, and computes the phase difference for each frequency. 
     The stencil filtering process unit  83  performs a stencil filtering process in Step S 63 . In other words, a stencil filtering process of accumulating phase difference patterns periodically appearing in each frequency of the sound collecting signals is performed. 
     The sound source direction estimation unit  84  estimates the direction from which a sound comes in Step S 64 . In other words, among phase-frequency patterns obtained by performing the stencil filtering process on all phase differences, patterns corresponding to frequency bands in which the position of a sound source is desired to be estimated are accumulated in a frequency direction, and then the direction from which the sound comes is estimated. The accumulation value becomes the maximum in the direction from which the sound comes. Thus, it is possible to estimate the direction from which the sound comes from the maximum accumulation value. 
       FIG. 12  is a diagram illustrating an example of a result of sound source direction estimation. In  FIG. 12 , the horizontal axis represents angles denoting the direction of a sound source, and the vertical axis represents accumulation values. In the example of  FIG. 12 , since the accumulation value is at the maximum in the direction of 20°, it is estimated that the virtual speaker  121  as a sound source is positioned in the direction of 20° with respect to the microphones  31 . 
     The sound source direction estimation unit  84  executes a process of generating an equation in Step S 65 . In other words, the sound source direction estimation unit  84  solves an equation of a function expressing a linear equation of a line passing through the center point P m  between the sound source and microphones  31  from an angle indicating the direction of the sound source. 
       FIG. 13  is a diagram illustrating an example of disposition of the microphones  31 .  FIG. 13  illustrates an example when sound source position is estimated on a two-dimensional plane. In  FIG. 13 , with the first microphone  31   a  as the center axis O, the second microphone  31   b  is disposed on the x axis, and the third microphone  31   c  is disposed on the y axis. In addition, the distance between the first microphone  31   a  and the second microphone  31   b  and the distance between the first microphone  31   a  and the third microphone  31   c  are set to be d. In other words, the microphones  31  are disposed so as to be positioned on the vertexes of a right-angle isosceles triangle. 
     Next, with reference to  FIGS. 14 to 16 , an example of a function of a linear equation of a line passing through the center point of an estimated sound source  201  and each microphone  31  is shown. 
       FIG. 14  is a diagram illustrating an example of a function of a linear equation of a line passing through the center point P m  between the estimated sound source  201  and the first microphone  31   a  and the second microphone  31   b . In  FIG. 14 , φ xo  is an angle that is formed by the perpendicular line to the line connecting the first microphone  31   a  and the second microphone  31   b  on the two-dimensional plane and a line L 1  of the linear equation, and specified by the sound source direction estimation unit  84 . The line L 1  in this case is expressed by the formula (4): 
     
       
         
           
             
               
                 
                   y 
                   = 
                   
                     
                       ( 
                       
                         x 
                         - 
                         
                           d 
                           2 
                         
                       
                       ) 
                     
                     ⁢ 
                     
                       1 
                       
                         tan 
                         ⁢ 
                         
                             
                         
                         ⁢ 
                         
                           ϕ 
                           xo 
                         
                       
                     
                   
                 
               
               
                 
                   ( 
                   4 
                   ) 
                 
               
             
           
         
       
     
       FIG. 15  is a diagram illustrating an example of a function of the linear equation of a line passing through the center point P m  between the estimated sound source  201  and the first microphone  31   a  and the third microphone  31   c . In  FIG. 15 , φ yo  is an angle that is formed by the perpendicular line to the line connecting the first microphone  31   a  and the third microphone  31   c  on the two-dimensional plane and a line L 2  of the linear equation, and specified by the sound source direction estimation unit  84 . The line L 2  in this case is expressed by the formula (5): 
     
       
         
           
             
               
                 
                   y 
                   = 
                   
                     
                       ( 
                       
                         x 
                         - 
                         
                           d 
                           2 
                         
                       
                       ) 
                     
                     ⁢ 
                     
                       1 
                       
                         tan 
                         ⁢ 
                         
                             
                         
                         ⁢ 
                         
                           ϕ 
                           yo 
                         
                       
                     
                   
                 
               
               
                 
                   ( 
                   5 
                   ) 
                 
               
             
           
         
       
     
       FIG. 16  is a diagram illustrating an example of a function of the linear equation of a line passing through the center point P m  between the estimated sound source  201  and the second microphone  31   b  and the third microphone  31   c . In  FIG. 16 , φ xy  is an angle that is formed by the perpendicular line to the line connecting the second microphone  31   b  and the third microphone  31   c  on the two-dimensional plane and a line L 3  of the linear equation, and specified by the sound source direction estimation unit  84 . The line L 3  in this case is expressed by the formula (6): 
     
       
         
           
             
               
                 
                   
                     ( 
                     
                       y 
                       - 
                       
                         d 
                         2 
                       
                     
                     ) 
                   
                   = 
                   
                     
                       ( 
                       
                         x 
                         - 
                         
                           d 
                           2 
                         
                       
                       ) 
                     
                     ⁢ 
                     
                       tan 
                       ⁡ 
                       
                         ( 
                         
                           
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                             4 
                           
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                         ) 
                       
                     
                   
                 
               
               
                 
                   ( 
                   6 
                   ) 
                 
               
             
           
         
       
     
     When more number of microphones  31  are used, it is preferable to perform the processes of Steps S 61  to S 65  on all pairs of microphones. In the same manner, by defining the center points between the microphones  31  included in specific pairs, and the angle that is formed by the perpendicular lines to the line connecting the microphones  31  included in the specific pairs on a two-dimensional plane and a line of the linear equation in a three-dimensional space, the respective elements are computed. 
     Next, the position detection unit  85  executes a process of solving the simultaneous equations in Step S 66 . When there is no solution to the simultaneous equations, a least-squares error solution is obtained. Then, the position detection unit  85  detects the positions of the microphones  31  with respect to the standard reference listening point P 0  in Step S 67 . In other words, by solving the simultaneous equations, a vector V 12  (refer to  FIG. 9 ) as position information indicating the position of the virtual speaker  121  as the sound source  201  with respect to the microphones  31  is obtained. 
     The position of the virtual speaker  121  is also included on the sound field correction data map stored in the non-volatile memory  28 . As illustrated in  FIG. 9 , the position of the virtual speaker  121  stored on the sound field correction data map is a vector V 11  as position information indicating the position with respect to the standard reference listening point P 0 . Thus, the position difference data V 1  that is a vector indicating the position of a microphone  31  with respect to the standard reference listening point P 0  is obtained from the vector V 11  and the vector V 12 . As described above, vector V 11  is a vector indicating the position of the virtual speaker  121  with respect to the standard reference listening point P 0 , and the vector V 12  is a vector obtained by solving the simultaneous equations, indicating the position of the virtual speaker  121  with respect to the microphone  31 . 
     Next, with reference to  FIG. 17 , a specific example of the position detection process on the two-dimensional plane will be further described.  FIG. 17  is a flowchart showing the position detection process on the two-dimensional plane. First, signals S x (n), S y (n), and S o (n) of n (in the case of the drawing, n=3) microphones  31   a ,  31   b , and  31   c  shown in  FIGS. 14 to 16  are acquired (Steps S 61   b , S 61   c , and S 61   a ). Sound source directions φ xo , φ yo , and φ xy  are estimated from the pairs of signals using stencil filtering (Steps S 64   a , S 64   b , and S 64   c ). To be specific, the sound source direction φ xo  is estimated using the signals S x (n), and S o (n), the sound source direction φ yo  is estimated using the signals S y (n) and S o (n), and the sound source direction φ xy  is estimated using the signals S x (n) and S y (n). 
     The following simultaneous equations including linear equations of lines indicating the sound source directions are generated (Step S 65 ). 
     
       
         
           
             
               
                 
                   
                     
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     The least-squares error solutions to the generated simultaneous equations are obtained (Step S 66 ). 
     
       
         
           
             
               
                 
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                   = 
                   
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     From the solutions, the sound source coordinates (x, y) which are coordinates of the virtual speaker  121  are estimated, and further, the position of the microphones  31  are detected from the sound source coordinates (x, y) (Step S 67 ). 
     In the above description, the position detection process is executed by the CPU  21 , but can be executed by the DSP  22 . 
     Modification Example 
     In Step S 32  of  FIG. 5 , the average value of the parameters for frequency characteristic correction of left channels and right channels is set to be computed, however, the following process can still be executed. In other words, any one of the parameters for frequency characteristic correction of the right channels and the left channels can be selected based on the size of the amount of frequency correction of right channels and the left channels. 
     To be specific, an additional value of the absolute values of the gains G of the left channels of G LS =|G L1 |+|G L2 |+|G L3 |+ . . . and an additional value of the absolute values of the gains G of the right channels of G RS =|G R1 |+|G R2 |+|G R3 |+ . . . are computed. Then the additional values of G LS  and G RS  are compared, and among the additional values, a parameter for frequency correction of a smaller value or a greater value than the other, in other words, any one of the right channels and the left channels is set in the equalizer devices  62   1  and  62   2 . 
     In addition, instead of computing the average value of the parameters for frequency correction of the right channels and the left channels, any one of the parameters for frequency correction of the right channels and the left channels can also be selected based on the size of the amount of the frequency correction in a predetermined frequency band. 
     To be specific, the sizes of a gain G L  of a left channel and a gain G R  of a right channel in a predetermined frequency band Δf are determined. Then, among the gains a parameter for frequency correction of a smaller gain or a greater gain than the other, in other words, any one of the right channel and the left channel is set in the equalizer devices  62   1  and  62   2 . 
     Furthermore, instead of computing the average value of the parameters for frequency correction of the right channels and the left channels, the sizes of the frequency correction amount are determined for predetermined frequency bands, and a parameter for frequency correction of a greater amount or a smaller amount of the correction than the other in a frequency band can also be selected. 
     To be specific, the sizes of a gain G L  of a left channel and a gain G R  of a right channel in each frequency band Δf are determined, and then among the gains, a parameter for frequency correction of a smaller gain or a greater gain than the other is set in the equalizer devices  62   1  and  62   2 . 
     Furthermore, instead of computing the average value of the parameters for frequency correction of the right channels and the left channels, the following process can also be executed. In other words, the difference between the amount of frequency correction of the right channels and the amount of frequency correction of the left channels are computed. Then, when the difference of the amounts of correction is equal to or higher than a predetermined value, a correction process of a frequency characteristic is performed on sound signals of the right channels and the left channels. When the difference of the amounts of correction is less than the predetermined value, the correction process of the frequency characteristic is not performed on the sound signals of the right channels and the left channels. 
     To be specific, an additional value of the absolute values of the gains G of the left channels of G LS =|G L1 |+|G L2 |+|G L3 |+ . . . and an additional value of the absolute values of the gains G of the right channels of G RS =|G R1 |+|G R2 |+|G R3 |+ . . . are computed. Then, the difference between the additional value G LS  and the additional value G RS |G LS −G RS | is computed, and then it is determined whether or not the difference |G LS −G RS | is equal to or greater than the predetermined value ΔG. When the difference |G LS −G RS | is equal to or greater than ΔG, the average value or any one of the parameters for frequency correction of the right channels and the left channels is set in the equalizer devices  62   1  and  62   2 . When difference |G LS −G RS | is less than ΔG, the correction process of the frequency characteristic is not performed on the sound signals of the right channels and left channels. 
     In addition, the following operation can also be performed. In other words, the difference between the gain G L  of the left channel and the gain G R  of the right channel, which is |G L −G R | in a predetermined frequency band Δf is computed. It is determined whether or not the difference |G L −G R | is equal to or greater than a predetermined value ΔG. When the difference |G L −G R | is equal to or greater than ΔG, the average value or any one of the parameters for frequency correction of the right channels and the left channels is set in the equalizer devices  62   1  and  62   2 . On the other hand, when the difference |G L −G R | is less than ΔG, the correction process of the frequency characteristic is not performed on the sound signals of the right channels and left channels. 
     Furthermore, the following operation can also be performed. In other words, the difference between the gain G L  of the left channel and the gain G R  of the right channel, which is |G L −G R | in each frequency band Δf is computed, and then it is determined whether or not the difference |G L −G R | is equal to or greater than the predetermined value ΔG. When the difference |G L −G R | is equal to or greater than ΔG, the average value or any one of the parameters for frequency correction of the right channels and the left channels is set in the equalizer devices  62   1  and  62   2 . When the difference |G L −G R | is less than ΔG, the correction process of the frequency characteristic is not performed on the sound signals of the right channels and left channels. 
     The above-described values are merely examples, and different values may also be used if necessary. In addition, each of configurations described above can be combined with each other as long as it does not depart from the gist of the present technology. 
     Furthermore, in the above-described embodiment, the case in which the correction process of a frequency characteristic is performed on sound signals of the front-left channel and the front-right channel has been described, but the following configuration can also be employed. In other words, the correction process of frequency characteristics may also be performed on sound signals of the surround-left channel, the surround-right channel, the surround-back left channel, the surround-back right channel, and the like. 
     Note that, sound signals on which the correction process of the frequency characteristics is performed are arbitrary. It is preferable to set the signals to be sound signals of channels output from a side or the back side of a user, such as the surround-left channel, the surround-right channel, the surround-back left channel, the surround-back right channel, and the like. This is because little shift in phases of the sound signals of the surround-left channel, the surround-right channel, the surround-back left channel, the surround-back right channel, and the like is preferable in order to obtain an appropriate surround effect. 
     In addition, in the embodiment of  FIG. 3 , one correction unit  71  is provided, but as shown in  FIG. 18 , two correction units can also be provided.  FIG. 18  is a block diagram showing another configuration example of the correction unit  71 . In the embodiment of  FIG. 18 , the correction unit  71  includes a first correction unit  301  and a second correction unit  302  that is subordinately connected to the first correction unit so that each can individually set parameters. 
     The first correction unit  301  and the second correction unit  302  have the same configuration. In other words, the first correction unit  301  includes the sound field processing device  61 , the equalizer device  62 , and the delay device  63  as the correction unit  71  of  FIG. 3 . The second correction unit  302  also includes the sound field processing device  61 , the equalizer device  62 , and the delay device  63  as the correction unit  71  of  FIG. 3 . 
     In the first correction unit  301 , the parameter PR 0  of the formula (3) is set as the first sound field correction data stored in the non-volatile memory  28  in Step S 41  of  FIG. 8 . In the second correction unit  302 , the sound field correction data D′ j  that is the second sound field correction data on the sound field correction data map  101  stored in the non-volatile memory  28  in advance is set. As a result, a sound field correction process using the parameter PR 0  is performed by the first correction unit  301 , a sound field correction process using the sound field correction data D′ j  is performed by the second correction unit  302 , and accordingly, a sound field correction process using the parameter PR j  of the formula (3) is collectively performed. When the listening point P k  is positioned on the standard reference listening point P 0 , sound field correction data D′ 0  becomes 0. Thus, the second correction unit  302  does not substantially perform a special process, and just outputs sound signals input from the first correction unit  301  to the later stage without change. Also in the embodiment, the same effect as in the embodiment of  FIG. 3  can be realized. 
     Application of the present technology to a program 
     A series of processes described above can be executed by hardware, or by software. 
     When the series of processes is executed by software, a program constituting the software is installed in a computer into which a dedicated hardware is incorporated or, for example, a general-purpose personal computer that can execute various functions from a network or a recording medium by installing various programs. 
     Such a recording medium in which the program is stored is configured as a removable medium  29  such as a magnetic disk (including a floppy disk), an optical disc (including a compact disk-read only memory (CD-ROM) or a DVD), a magneto-optical disc (including a mini-disc (MD)), or a semiconductor memory on which programs are recorded and is distributed to provide users with the programs separate from the main body of a device, and configured as a flash ROM or a hard disk included in a storage unit on which programs are recorded to provide users with them in a state of being incorporated into the main body of a device in advance. 
     Note that, in the present specification, steps of describing a program recorded in a recording medium include not only processes performed in time series in an order but also processes executed in parallel or an individual manner, not necessarily executed in a time series manner. 
     In addition, in the present specification, a system means a set of a plurality of constituent elements (devices, modules (components), and the like), and it does not matter whether or not all constituent elements are accommodated in the same housing. Thus, a plurality of devices which are accommodated in separate housings and connected with each other via a network, and one device of which one housing accommodates a plurality of modules are all systems. 
     Note that, embodiments of the present technology are not limited to the above, and can be variously modified within the scope not departing from the gist of the present technology. 
     Other Configurations 
     The present technology can also employ the following configurations: 
     (1) A sound processing device which includes a shift detection unit that detects a shift of a listening point on which a user listens to a sound in a sound field space from a standard reference listening point, and a correction unit that corrects a sound signal based on first sound field correction data for correcting a sound field when the listening point is the standard reference listening point and second sound field correction data for correcting a sound field of the listening point when the listening point is shifted from the standard reference listening point. 
     (2) The sound processing device described in (1) above, in which the second sound field correction data is data of a peripheral reference listening point in a position shifted from the standard reference listening point. 
     (3) The sound processing device described in (2) above, in which, when the position of the listening point is different from the position of the peripheral reference listening point, the second sound field correction data on the nearest peripheral reference listening point is the second sound field correction data of the listening point. 
     (4) The sound processing device described in any one of (1) to (3) above, in which the shift of the listening point from the standard reference listening point is detected by collecting a measurement sound generated from the listening point. 
     (5) The sound processing device described in (4) above, in which the shift of the listening point from the standard reference listening point is detected from the position of the listening point with respect to a position in which the measurement sound generated from the listening point is collected and a position in which the measurement sound is collected with respect to the standard reference listening point. 
     (6) The sound processing device described in (5) above, in which the position of the listening point with respect to the position in which the measurement sound generated from the listening point is collected is detected from a sound signal obtained by collecting the measurement sound generated from the listening point. 
     (7) The sound processing device described in (5) or (6) above, in which the position in which the measurement sound is collected with respect to the standard reference listening point is detected from position information of a sound emitting unit that emits a sound that the user listens to on the listening point in the sound field space with respect to the position in which the measurement sound is collected and position information of the sound emitting unit with respect to the standard reference listening point. 
     (8) The sound processing device described in any one of (1) to (7) above, which further includes a measurement unit that obtains the first sound field correction data by measuring a sound field of the sound field space. 
     (9) The sound processing device described in any one of (1) to (8) above, which further includes a storage unit that stores the second sound field correction data in advance. 
     (10) The sound processing device described in (9) above, in which the storage unit further stores the first sound field correction data measured by the measurement unit. 
     (11) The sound processing device described in any one of (1) to (10) above, in which a parameter calculated from the first sound field correction data and the second sound field correction data is set in the correction unit. 
     (12) The sound processing device described in any one of (1) to (10) above, in which the first sound field correction data and the second sound field correction data are set as individual parameters in different correction units. 
     (13) A sound processing method which includes steps of detecting a shift of a listening point on which a user listens to a sound in a sound field space from a standard reference listening point, and correcting a sound signal based on first sound field correction data for correcting a sound field when the listening point is the standard reference listening point and second sound field correction data for correcting a sound field of the listening point when the listening point is shifted from the standard reference listening point. 
     (14) A program that causes a computer to execute processes including steps of detecting a shift of a listening point on which a user listens to a sound in a sound field space from a standard reference listening point, and correcting a sound signal based on first sound field correction data for correcting a sound field when the listening point is the standard reference listening point and second sound field correction data for correcting a sound field of the listening point when the listening point is shifted from the standard reference listening point. 
     (15) A recording medium on which is recorded a program that causes a computer to execute processes including steps of detecting a shift of a listening point on which a user listens to a sound in a sound field space from a standard reference listening point, and correcting a sound signal based on first sound field correction data for correcting a sound field when the listening point is the standard reference listening point and second sound field correction data for correcting a sound field of the listening point when the listening point is shifted from the standard reference listening point. 
     The present disclosure contains subject matter related to that disclosed in Japanese Priority Patent Application JP 2012-219978 filed in the Japan Patent Office on Oct. 2, 2012, 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.