Patent Publication Number: US-2007110129-A1

Title: Method for measuring frequency characteristic and rising edge of impulse response, and sound field correcting apparatus

Description:
CROSS REFERENCES TO RELATED APPLICATIONS  
      The present invention contains subject matter related to Japanese Patent Application JP 2005-315738 filed in the Japanese Patent Office on Oct. 31, 2005, the entire contents of which are incorporated herein by reference.  
     BACKGROUND OF THE INVENTION  
      1. Field of the Invention  
      The present invention relates to a method for measuring a frequency characteristic and a rising edge of an impulse response, and a sound field correcting apparatus.  
      2. Description of the Related Art  
      With the growing popularity of DVDs (Digital Versatile Discs) and digital broadcasting, multi-channel audio systems have become increasingly widespread in general households. This has increased the need for listeners (users) to perform various setting and adjustment operations of audio channels by themselves.  
      However, setting and adjustment operations in multi-channel audio systems are complicated and are often very difficult for users who are not skilled in operation of this type of system. Thus, attempts have been made to enable devices in multi-channel audio systems such as AV amplifiers to perform correction processing in audio reproduction, so as to simplify or setting and adjustment to be performed by the users or omit the need for such setting and adjustment.  
      Such correction processing is referred to as, for example, “automatic sound field correction”, in which correction is performed on the basis of a result of measurement of an impulse response in a reproduction sound field. Specifically, the following processing procedure may be performed: (a) an impulse signal as shown in the left side of  FIG. 14A  is supplied to a speaker of a channel of interest so that impulse sound is emitted; (b) the impulse sound is picked up by a microphone installed at a listening position of a user, and a signal representing an impulse response of a reproduction sound field (impulse response signal) as show in the right side of the  FIG. 14A  is obtained; (c) the impulse response signal is analyzed so that parameters for sound field correction are obtained; (d) an audio signal of the channel of interest is corrected using the parameters for sound field correction.  
      However, the use of an impulse can degrade the S/N ratio of an output signal of a microphone. Thus, a technique has been developed in which an impulse is converted into pulse in which the energy of the impulse is dispersed in the time domain, and the converted pulse is used for sound field correction.  
      The pulse obtained through the conversion above is referred to as a “TSP (Time Stretched Pulse)”. An example of a waveform of a TSP signal is shown in the left side of  FIG. 14B , in which N represents the length of a TSP, indicating the total number of samples in one TSP signal (N=4096, for example) and TN represents a period of N samples (unit period).  
      In this case, in order to convert an impulse into a TSP, the phase of pulse contained in the impulse is advanced in proportion to the square of the frequency. In order to convert the TSP back to the impulse, the phase of pulse contained in the TSP is retarded in proportion to the square of the frequency.  
      Specifically, the impulse is transformed using equations (1) and (2) shown in  FIG. 15 , so that the TSP in which the energy of the impulse is dispersed in the time domain can be obtained. The TSP is inversely transformed using equations (3) and (4) shown in  FIG. 15 , so that the dispersed energy is compressed and the impulse can be obtained again, as shown in the left side of  FIG. 14A  and the left side of  FIG. 14B .  
      Thus, when a TSP is employed, the following processing procedure can be performed: (e) the above processing of (a) and (b) is performed using a TSP signal instead of an impulse signal, so that a signal representing a TSP response in a reproduction sound field can be obtained as shown in the right side of  FIG. 14B ; (f) the dispersed energy in the TSP response signal is compressed again so that the TSP is inversely transformed to obtain an impulse response signal as shown in the right side of  FIG. 14A ; (g) the processing of (c) and (d) is performed using the impulse response signal.  
      With this TSP method, since the impulse energy is dispersed in the time domain, the S/N ratio of an output signal of a microphone is improved, and thus the accuracy of sound field correction is increased.  
       FIGS. 16A and 16B  are diagrams illustrating a timing in measurement of an impulse response using a TSP. As shown in the figures, a TSP signal comprises 4090 samples (N=4096) and is supplied to a speaker during each of periods T 1 , T 2 , . . . , and Tk. This indicates that a TSP response signal is output from a microphone with a delay of a period Td for each of the periods T 1 , T 2 , . . . , and Tk.  
      In this example, the length of each of the periods T 1  to Tk is same as that of a period TN. In the delay time Td, a leading period Ta corresponds to a distance between the speaker and the microphone, and a trailing period Ts corresponds to a system delay. Thus, the period Ta depends on the distance between the speaker and the microphone, and the period Ts has a predetermined value. The TSP response signals corresponding to the TSP signals are obtained k times. At this time, these TSP response signals are the same as each other.  
      Accordingly, when the TSP response signals are examined for each of the periods T 1 , T 2 , T 3 , . . . , and Tk, the TSP response signal obtained during the period T 2  can be considered as corresponding to the TSP supplied during the period T 2 . During this period T 2 , the first measurement of TSP response can be performed.  
      In addition, the TSP response signal obtained during the period T 3  is considered as corresponding to the TSP supplied during the period T 3 . Thus, the second measurement of TSP response can be performed during the period T 3 . Likewise, the TSP response signal obtained during the period Tk can be considered as corresponding to the TSP supplied during the period Tk. Thus, the (k−1)th measurement of TSP response can be performed during the period Tk.  
      However, it is not possible to consider that the TSP response signal obtained during the period T 1  is the TSP supplied during the period T 1 , since the TSP signal contains a noise signal representing the background noise. Thus, TSP response may not be measured during the period T 1 .  
      As can be seen from the foregoing, when TSP sound is continuously output k times, (k−1) TSP response signals can be obtained. These (k−1) TSP response signals are basically the same as each other and thus can be synchronously added together. At this time, the TSP response signals are averaged, and as a result influence of signal variance and noise is reduced to a negligible level.  
      Techniques related to the above technique are described in the following documents:  
      Nobuharu Aoshima, “Computer-generated pulse signal applied for sound measurement”, J. Acoust. Soc. Am., No. 69(5), May 1981;  
      Yoiti Suzuki, et al., “An optimum computer-generated pulse signal suitable for the measurement of very longimpulse responses”, J. Acoust. Soc. Am., No. 97(2), February 1995;  
      Yoiti Suzuki, et al., “Considerations on the design of time-stretched pulses”, Technical Report of IEICE, EA92-86(1992-12);  
      Hutoshi Asano “Measuring impulse response using TSP”, RWCP Sound Scene Database in Real Acoustical Environments, 5 Feb. 2001, available from http://tosa.mri.co.jp/sounddb/tsp/tsp_circular.htm.  
      According to the above documents or other documents, in measuring an impulse response using a TSP, the length N (the number of samples) of the TSP needs to be greater than that (the number of samples) of the corresponding impulse response (i.e., a period lasting until the effective amplitude become sufficiently small), as shown in  FIGS. 14A and 14B . Thus, the following equation has to be satisfied.
 
N&gt;ν  (5)
 
      This can also be understood from  FIG. 16 . As illustrated in the figure, when the effective period of the TSP response signal increases, an effective portion or “tail” of a TSP response signal is superimposed around the leading part of the subsequent TSP response signal.  
      Thus, for example, when the sampling frequency of the TSP is 48 kHz and the reverberation time is 0.5 seconds, the length N of the TSP will be greater than 24000 samples (=0.5 seconds). When a FFT (fast Fourier transform) technique is used for inversely transforming a TSP, the length N is equal to a power of two and thus the length N is 32768 in this example.  
      If a sound field has a longer reverberation time which is based on the size of a room and reflecting objects, the length of an impulse, a longer impulse response (ν) is obtained and thus a longer TSP, which is corresponding to the impulse response, will be obtained. This disadvantageously causes an increase in measuring time and in resources necessary in the measurement such as a CPU, a DSP, and memory.  
     SUMMARY OF THE INVENTION  
      The present invention has been made in view of the above circumstances.  
      Accordingly, in a method for measuring a frequency characteristic of a sound field between a sound source and a receiver, N denotes the length of a TSP signal, ν denotes the length of an impulse response between the sound source and the receiver, TN denotes a duration period of the TSP signal, and T 1  to T(k+L) denote periods each composed of the period TN as a unit period (k≧1, L≧0). This method includes the steps of setting N so as to satisfy N≦ν, supplying the TSP signal to the sound source continuously for each unit period TN over the periods T 1  to Tk, adding and averaging signals output from the receiver during the individual periods T 1  to T(k+L), and performing circular convolution on a value obtained by the adding and averaging so that the frequency characteristic of the sound field between the sound source and the receiver is obtained.  
      According to an embodiment of the present invention, TSP sound is continuously output to a reproduction sound field, and addition/averaging and circular calculation is performed on corresponding TSP response signals. This permits a decrease in the time necessary for measurement of impulse response and a reduction in resources necessary for the measurement such as a CPU, a DSP, and a memory. 
    
    
     BRIEF DESCRIPTION OF THE DRAWINGS  
       FIGS. 1A  to  1 D illustrate a timing diagram illustrating an embodiment of the present invention;  
       FIG. 2  illustrates an embodiment of the present invention;  
       FIGS. 3A and 3B  are waveform diagrams illustrating an embodiment of the present invention;  
       FIGS. 4A and 4B  are waveform diagrams illustrating an embodiment of the present invention;  
       FIGS. 5A and 5B  are waveform diagrams illustrating an embodiment of the present invention;  
       FIG. 6  illustrates an embodiment of the present invention;  
       FIG. 7  illustrates an embodiment of the present invention;  
       FIG. 8  is a flowchart illustrating signal processing according to an embodiment of the present invention;  
       FIG. 9  is a flowchart illustrating signal processing according to an embodiment of the present invention;  
       FIGS. 10A  to  10 C are waveform diagrams illustrating an embodiment of the present invention;  
       FIG. 11  is a waveform diagram illustrating an embodiment of the present invention;  
       FIG. 12  is a characteristic diagram illustrating an embodiment of the present invention;  
       FIG. 13  is a block diagram illustrating a system according to an embodiment of the present invention;  
       FIGS. 14A and 14B  are waveform diagrams illustrating a TSP signal;  
       FIG. 15  illustrates a TSP signal; and  
       FIGS. 16A and 16B  are waveform diagrams illustrating a TSP signal. 
    
    
     DESCRIPTION OF THE PREFERRED EMBODIMENTS  
      [1] Concept of the Present Invention  
      In known measurement techniques, as described in the above documents or other documents, a TSP used for impulse response measurement is prepared after it has been verified that the length N of the TSP as an output and the length ν of an impulse response in a reproduction sound field can satisfy the above equation (5).  
      However, for a normal sound field, it is sufficient to correct the frequency response characteristics and time alignment (time delay correction), and thus only acquisition of parameters for the correction is necessary.  
      Accordingly, the present embodiment is intended not for “accurate calculation of an impulse response” but for “accurate derivation of parameters for correction of a sound field”. Thus, a TSP which is shorter than a reverberation time as expressed by the following equation is used.
 
N≦ν  (6)
 
 Specifically, TSP sound corresponding to such a TSP is continuously output to a reproduction sound field so that parameters used for sound field correction can be obtained through addition/averaging and circular calculation. This allows reduction of a measuring time as well as resources used in the measurement such as a CPU, a DSP, and a memory. 
 
 [2] Frequency Characteristic (Frequency Amplitude Characteristic) 
 
      A frequency characteristic can be obtained by performing adequate synchronous addition, even when the values of N and ν have a relationship expressed by the equation (6). This will be described in more detail below.  
      [2-1] TSP Response Signal  
       FIGS. 1A  to  1 D shows a timing diagram illustrating measurement of a TSP response using a TSP. As show in  FIG. 1A , a TSP signal is composed of 4096 samples (N=4096). This TSP signal is continuously supplied to a speaker during each of periods T 1 , T 2 , . . . , and Tk. In this embodiment, the value of k is assumed to be ten, similarly to the case described using  FIG. 16 .  
      Thus, as shown in  FIG. 16B , a TSP response signal SR 1  is obtained from TSP sound emitted in the period T 1 . FIG.  1  illustrates a case where one TSP response signal is obtained over a four-unit period TN. Thus, the TSP response signal SR 1  is obtained over the periods T 1  through T 4  with a delay of a period Td from the start point of the period T 1 .  
      When a sampling frequency of the TSP signal output as TSP sound is 48 kHz, the unit period TN is 4096/48000≈85.3 [ms]. When the velocity of sound in air is 340 m/s, the propagation distance of an acoustic wave is 340 [m/s]×85.3 [ms]≈29 [m]. Thus, in an ordinary room in which AV (Audio/Visual) reproduction is performed, Td&lt;TN is satisfied, and the head of the TSP response signal SR 1  is within the period T 1  (i.e., the TSP response signal SR 1  is output during the period T 1 ).  
      A TSP response signal SR 2  is obtained from TSP sound emitted over the periods T 2  to T 5 . Likewise, from TSP sound emitted in a period T 1  (i=1 to k), a TSP response signal Sri is obtained over periods T 1  to T (i+3).  
      Then, as shown in  FIG. 1 , in TSP response signal SR 1 , a signal component corresponding to the period T 1  is assumed to be a signal S 1 , and likewise, a signal component corresponding to the period T 2  to be S 2 , a signal component corresponding to the period T 3  to be S 3 , a signal component corresponding to the period T 3  to be T 4 .  
      The subsequent TSP response signal SR 2  is basically the same as the signal SR 1  except that the signal SR 2  is shifted by the unit period TN from the signal SR 1 . Thus, in the TSP response signal SR 2 , a signal component corresponding to the period T 2  can be regarded as the signal S 1 , a signal component corresponding to the period T 3  as the signal S 2 , a signal component corresponding to the period T 4  as the signal S 3 , and a signal component corresponding to the period T 5  as the signal S 4 .  
      Likewise, since the TSP response signals SR 1  to SRk are the same as each other except that the start point of each of these signals are shifted by TN, for any TSP response signal SR 1 , a signal component corresponding to a period Ti can be regarded as the signal S 1 , and a signal component corresponding to a period T (i+3) can be regarded as the signal S 4 .  
      In the actual situation, an output signal from a microphone is a signal composed of the signals SR 1  to SRk added together. Thus, as shown in  FIG. 1C , the signal S 1  is obtained in the period T 1 , a signal (S 1 +S 2 ) is obtained in the period T 2 , a signal (S 1 +S 2 +S 3 ) is obtained in the period T 3 , and a signal (S 1 +S 2 +S 3 +S 4 ) is obtained in the period T 4 . Similarly, in each of the periods T 5  to Tk, the signal (Si+S 2 +S 3 +S 4 ) is obtained.  
      Since no TSP sound is emitted during periods T(k+1) to T (k+3), a signal (S 2 +S 3 +S 4 ) is obtained in the period T(k+1), a signal (S 3 +S 4 ) is obtained in the period T (k+2), and the signal S 4  is obtained in the period T (k+3).  
      Then, as shown in the bottom of  FIG. 1D , the individual signals obtained during the periods T 1  to T (k+3) are added together as follows:  
                 S   ⁢           ⁢   1     +     (       S   ⁢           ⁢   1     +     S   ⁢           ⁢   2       )     +     (       S   ⁢           ⁢   1     +     S   ⁢           ⁢   2     +     S   ⁢           ⁢   3       )     +       (       S   ⁢           ⁢   1     +     S   ⁢           ⁢   2     +     S   ⁢           ⁢   3     +     S   ⁢           ⁢   4       )     ×     (     k   -   3     )       +     (       S   ⁢           ⁢   2     +     S   ⁢           ⁢   3     +     S   ⁢           ⁢   4       )     +     (       S   ⁢           ⁢   3     +     S   ⁢           ⁢   4       )     +     S   ⁢           ⁢   4       =         (       S   ⁢           ⁢   1     +     S   ⁢           ⁢   2     +     S   ⁢           ⁢   3     +     S   ⁢           ⁢   4       )     ×   k     ≡     k   ·   SW               (   7   )             
 
      Specifically, the TSP response signals SR 1  to SRk obtained during the periods T 1  to T (k+3) are divided with respect to each unit period TN, and the signals obtained in the individual periods T 1  to T (k+3) are added together. The result is divided by the number k of TSP sound emissions so as to be averaged. Consequently, as shown in  FIG. 2 , a signal Sw is obtained which is composed of the signals S 1  to S 4  of the TSP response signal SR 1  for each N-sample period TN.  
      In a general way, when TSP sound is emitted k times during the periods T 1  to Tk, response signals corresponding to the TSP sound are measured (k+L) times for each N-sample period TN during the period T 1  to T (k+L). Then the response signals are added and averaged so that the signal Sw is obtained. The value L is the number of no-sound periods, subsequent to the period Tk, during which TSP response sound is picked up, which will be described in detail below.  
      If a sufficiently acceptable result of frequency analysis performed on the signal Sw can be obtained as the frequency characteristic, the signal Sw can be used for deriving a parameter for sound field correction. This is described below. The Sw signal, which has been obtained through addition and averaging of the TSP response signals SR 1  to SRk for each unit period TN, is hereinafter referred to as a “wrapped signal”, and the addition/averaging processing for the wrapped signal is hereinafter referred to as “wrapping processing”.  
      [2-2] Comparison Between Characteristics of Impulse Response Signal and Wrapped Signal  
      In the following, waveform characteristics of an impulse response signal and a generated wrapped signal will be described.  FIG. 3A  illustrates an example of a waveform of an impulse response signal with 1024 samples, and  FIG. 3B  illustrates a waveform showing amplitude values obtained by performing an FFT on the impulse response signal.  FIG. 4A  illustrates a waveform of a wrapped signal which has undergone wrapping processing for each N-sample period (N=256), as described using  FIGS. 1 and 2 .  FIG. 4B  illustrates a waveform showing amplitude values obtained by performing an FFT on the wrapped signal. Note that the X-axes of  FIG. 3  and  FIG. 4  have different pitches (scales).  
      It can be seen that the waveform showing the FFT amplitudes shown in  FIG. 3B  and the waveform showing the FFT amplitudes shown in  FIG. 4B  resemble each other in general shape.  
       FIG. 5A  illustrates a leading part of the FFTed impulse signal (i.e., a leading part of the waveform illustrating the FFT amplitudes shown in  FIG. 3B ).  FIG. 5B  illustrates a leading part of the FFTed impulse signal (i.e., a leading part of the FFTed wrapped signal shown in  FIG. 4B ). As can be seen from  FIGS. 5A and 5B , the FFT amplitudes of the wrapped signal and the FFT amplitudes of the impulse response signal agree every four samples.  
      This agreement can be analyzed using formulas shown in  FIGS. 6 and 7 . As can be seen from the set of formulas, it is proved that a resultant FFT output of the wrapped signal is a part of a resultant FFT output of the impulse response signal.  
      Therefore, although a resolution obtained by analyzing the waveform of the wrapped signal by an FFT technique is lower than that obtained by analyzing the waveform of the impulse response signal by an FFT technique. However, in the frequency domain, the analysis of the waveform of the wrapped signal and the analyzing the waveform of the impulse response signal result in the same values. Accordingly, even in the case where the equation (6) is satisfied, i.e., where an impulse response signal in the actual sound field lasts for a period of 4096 or more samples, an accurate frequency characteristic can be measured using a wrapped signal, and thus an appropriate parameter for sound field correction can be obtained.  
      [2-3] Value L  
      When a frequency characteristic is measured using a TSP, as described above, the value L is set in accordance with the impulse response in a sound field of interest. This allows precise measurement of a frequency characteristic even in the case where the equation (6) is satisfied.  
      However, in order to set the value L to be associated with an impulse response, it is necessary to obtain the reverberation time of the sound field in advance. In this case, an increase in the value L means an increase in the sound pickup period T (k+1) or a later period (i.e., in a period during which no TSP sound is emitted). This indicates that noise signals representing the background noise are repeatedly added until the TSP response signal becomes sufficiently small. In addition, when the value L is set to be a fixed value, the pickup period may be unnecessarily long for a sound field with a short reverberation time, resulting in an increase in the measuring time.  
      Accordingly, in view of the S/N ratio and measuring time, it is desirable that L is decreased for a sound field with a short reverberation time and increased for a sound field with a long reverberation time.  
      The variable m of the equations (1) and (3) is a parameter associated with the length N of a TSP. However, the value m is not determined by the length ν of an impulse response. Thus, by setting the value m to be a value close to 2, a large phase rotation of a TSP signal can be obtained, resulting in a decrease in the amplitude of the TSP signal. Consequently, the gain of a measuring signal can be increased, which permits efficient measurement in terms of the S/N ratio.  
      [2-4] Example of Determination Scheme of Value L  
       FIGS. 8 and 9  show examples of algorithms for determining the value L. In each of these algorithms, the following processing procedure is performed: A. The magnitude of the background noise is measured in a preliminary period; B. Processing for the periods T 1  to Tk is performed; C. On the basis of the maximum value or the average value of the background noise as a reference value, the level of a picked-up response signal is checked in real time for each period TN after the period T (k+1) and thereafter; D. On the basis of the result of the check, whether the processing is continued or terminated is determined.  
      [2-4-1] When the Maximum Value is Used  
      The maximum value can be used in the algorithm shown in  FIG. 8 , in which the last period T (k+3) is determined on the basis of the maximum values of a background noise signal and a picked-up response signal. Specifically, in a routine  100  illustrated in the  FIG. 8 , processing is initiated at STEP  101  in response to an instruction of measurement of a frequency characteristic. At STEP  102 , the background noise is picked up for a predetermined period TN×M (M is a natural number). At STEP  103 , the maximum amplitude value MAX_noise of the picked-up signal is calculated.  
      At STEP  104 , TSP sound is emitted during the periods T 1  to Tk, as described using  FIG. 1 . At the same time, TSP responses corresponding to the TSP sound is picked up during the periods T 2  to Tk, and the TSP response signals are added together for each unit period TN so that a wrapped signal Sw is generated for each of the periods T 2  to Tk, as described with reference to  FIG. 1 . Then, at STEP  105 , the maximum amplitude value MAX_resp of the wrapped signal Sw is calculated.  
      At STEP  111 , as shown in  FIG. 1 , no TSP sound is output during the subsequent period T(k+1) (L=1). However, TSP response sound is picked up, and the maximum amplitude value MAX_tail of the picked-up TSP response signal (i.e., the maximum amplitude value in the period T (k+1)) is calculated. At STEP  112 , the maximum amplitude value MAX_tail is compared with a value (α·MAX_noise) obtained by multiplying the maximum amplitude value MAX_noise of the background noise calculated in STEP  103  by a predetermined magnification factor α (α&gt;1).  
      In the above comparison, if MAX_tail&gt;(α·MAX_noise), indicating that TSP response has been obtained, the processing procedure proceeds to STEP  114 . At STEP  114 , the TSP response signal picked up in the period T(k+1) in STEP  111  is added to the wrapped signal Sw corresponding to the periods T 2  to Tk and the resultant value is averaged. Then, the procedure returns to STEP  111 . At this time, the wrapped signal Sw is formed by adding and averaging the TSP response signals corresponding to the periods T 2  to T(k+1).  
      Thereafter, the processing of STEP  111  to STEP  114  are repeated for each of the periods T (k+2) and T (k+3). Consequently, the wrapped signal Sw is a signal formed by adding and averaging the TSP response signals corresponding to the periods T 2  to T (k+3).  
      In the period T (k+4), no TSP response is output, and only the background noise is present. At this time, since MAX_tail≦(α·MAX_noise), the procedure proceeds to STEP  113 . AT STEP  113 , the maximum amplitude value MAX_tail calculated in STEP  111  is compared with a value (β·MAX_resp) obtained by multiplying the maximum amplitude value MAX_resp of the wrapped signal Sw calculated in STEP  105  by a predetermined magnification factor (0&lt;β≦1).  
      In the above comparison, if MAX tail≦(β·MAX_resp), indicating that no TSP response has been obtained, the procedure proceeds to STEP  300 . At this time, the wrapped signal Sw is formed by adding and averaging the TSP response signals corresponding to the periods T 2  to T (k+3). Thus, frequency analysis or the like can be performed on the wrapped signal Sw so that a parameter used for sound field correction can be obtained.  
      On the other hand, if MAX_tail&gt;(β·MAX_resp), indicating that the TSP response is still present, the procedure proceed to STEP  114  and then returns to STEP  111 . Specifically, the termination of the TSP response signal is checked in both STEP  112  and STEP  113 , and the TSP response signal is determined to have been terminated in both of STEP  112  and STEP  113 , the wrapped signal Sw is analyzed and used for obtaining a parameter for sound field correction such as correction of the frequency characteristic.  
      Thus, according to the routine  100 , a wrapped signal Sw corresponding to TSP response signals can be appropriately obtained. This allows generation of a parameter for correction of the frequency characteristic.  
      [2-4-2] When Average Value is Used  
      The average value can be used in the algorithm shown in  FIG. 9 , in which the last period T (k+3) is determined on the basis of the average energy values of a background noise signal and a picked-up response signal. This processing is realized by a routine  200  illustrated in  FIG. 9 . Processing procedure in this routine  200  is similar to that in the routine  100 , the description of thereof is omitted. The reference numerals assigned to each processing of the routing  200  are different from those assigned to the corresponding processing of the routine  100 . In addition, in the routine  200 , “Eng_noise” denotes the average energy of the TSP response signal, “Eng_resp” denotes the average energy of the wrapped signal Sw, and “Eng_tail” denotes the average energy of the TSP response signal for each period TN of the period T (k+1) and later periods.  
      Also in this routine  200 , a wrapped signal Sw can be appropriately obtained, and thus a adequate parameter for correction of the frequency characteristic can be generated.  
      [2-4-3] Supplementary Explanation  
       FIG. 10A  illustrates an example of measurement in which an impulse response is measured over a 65536-sample period. As can be seen from the figure, the energy of the impulse response is concentrated in the initial 4096-sample period T 1  (=TN) when emitted within an expected range, and energy in the subsequent periods is significantly decreased.  
      Basically a TSP can be considered as being composed of an impulse train in different time instances. Therefore, the energy of the leading pulse contained in the TSP is concentrated in the initial period T 1  in a corresponding TSP response signal. Likewise, the energy of the trailing pulse contained in the TSP is concentrated in the subsequent 4096-sample period T 2  in the TSP response signal. In addition, as shown in the  FIG. 1 , k TSP response signals SR 1  to SRk are added and averaged so that the wrapped signal Sw is generated.  
      Thus, as the value k increases, influence of the value L on the wrapped signal Sw decreases. Even when the value L is a fixed value, error in the wrapped signal Sw can be reduced. For example, if the value of k is 32, the appropriate wrapped signal Sw can be obtained even where L=0. In this case, the processing described above in which the level of the TSP response signal is checked using the maximum value or average value of the background noise is not necessary, and thus the entire processing procedure can be simplified.  
      [3] Time Alignment  
      In the following, a measuring method intended for time alignment when the values N and ν has the relationship expressed by the equation (6) will be described.  
      [3-1] Rising Edge of Impulse Response  
      In time alignment, a parameter necessary for sound field correction is a distance between a sound source such as a speaker and a receiver such as a microphone. The distance corresponds to the time Ta (i.e., a time period obtained by subtracting the system delay time Ts from the delay time Td), as described using  FIG. 16 . Therefore, an impulse response signal is acquired from the wrapped signal Sw, and a rising edge of the impulse response signal can be analyzed.  
      As described in the foregoing, an impulse response is acquired through inverse TSP processing as expressed be the equations (3) and (4) in circular convolution using DFT or FFT which is performed on a TSP response signal (shown in  FIG. 1 ) obtained through continuous emission of TSP sound. However, the signal obtained through this technique is not an impulse response in a precise sense, but an impulse response which has undergone wrapping processing.  
      This may bring about a problem described below. As described above,  FIG. 10A  illustrate an example of measurement of an impulse response waveform.  FIG. 10B  shows an enlarged representation of the initial 4096-sample period T 1  in the time domain.  FIG. 10C  illustrates a waveform of an impulse response obtained by performing inverse TSP filtering on a wrapped signal Sw. This wrapped signal Sw is generated under the same condition as that under which the impulse response waveform is obtained, by performing addition and averaging of TSP response signals for each 4096-sample period. This inverse TSP-filtered waveform is also shown in  FIG. 10C  as an enlarged representation of the initial 4096-sample period T 1  in the time domain.  
      In each of  FIGS. 10B and 10C , a large amplitude change observed in the vicinity of 600 samples represents the initial rise caused by an impulse or TSP, and a period between the head of the waveform and the initial rise corresponds to the delay period Td. In the case of  FIG. 10B  (actual impulse response signal), only a noise component representing the background noise is present during the period Td between the head of the waveform and the initial rise. Therefore, the signal level is sufficiently small, allowing an initial rise point (rising edge) to be distinguished.  
      Thus, in this case, the rising edge can be detected by setting a threshold level V TH , by multiplying the maximum amplitude value of the impulse response signal by a predetermined ratio “a” (for example, “a”=20%). Then, a time point at which the impulse response signal exceeds the threshold level V TH  can be considered as the rising edge of the impulse response.  
      On the other hand, in the case of the  FIG. 10C  (impulse response signal obtained by inversely TSP-transforming the wrapped signal Sw (inverse-TSP impulse response signal)), signal components corresponding to the period T 2  and later periods are added to the signal component corresponding to the initial period T 1  for each sample period TN (N=4096). Thus, in the period Td, the signal components of the TSP response signal corresponding to the period T 2  and later periods are present. As a result, the signal component of the inverse-TSP impulse response signal in the period Td has a certain amplitude, which reduces the distinguishability of the rising edge of the impulse response as compared with the impulse response signal of  FIG. 1O B.  
      In setting of the threshold level V TH , if a large ratio “a” to be multiplied with the maximum amplitude value is set, a high threshold level V TH  is obtained. This decreases precision in time for detecting the rising edge of a waveform. On the other hand, however, a small ratio “a” results in a low threshold level V TH , which increases possibility of error in the detection of the rising edge. Specifically, amplitude fluctuation which occurs prior to the actual rise of the impulse response may be misrecognized as representing the rise of the impulse response.  
      Therefore, in the case of the inverse-TSP impulse response signal, it is not possible to set the threshold level V HT  by multiplying the maximum amplitude value of the impulse response signal by a predetermined ratio.  
      Accordingly, in this embodiment, the property described below is utilized so that the threshold level V TH  can be dynamically set.  
      [3-2] Measuring Method of Rising Edge of Impulse Response  
      As described above, the inverse-TSP impulse response signal does not represent an actual impulse response in a precise sense. In the following, a property of the reverberation characteristic of a typical impulse response in the time domain is employed: (A) in a waveform of a typical impulse response signal, energy of a reverberation component is smaller than that of a rising edge component and an initial reflected sound component subsequent to the edge component. Thus, a waveform of an inversely TSP-transformed impulse response signal is not significantly different in general shape from the waveform of a typical impulse response signal. This can be seen from the waveforms illustrated in  FIGS. 10B and 10C , and the rising edge can be detected from the waveforms; (B) In an inverse-TSP impulse response signal, it is highly likely that a signal component in the period Td, which lasts from the head of the waveform to the rising edge, is a noise component representing the background noise or a reverberation component produced by wrapping processing. Thus, it is necessary to prepare an arrangement so that the amplitude in the period Td is not detected; (C) In general, the amplitude and energy of the reverberation component shows a generally simple decrease over time. For example, the amplitude of the impulse response waveform illustrated in  FIG. 10A  decreases along the time axis.  
      In addition, as can be seen from a waveform illustrated in  FIG. 11  (same as the waveform in  FIG. 10C ), also in the inverse-TSP impulse response signal, the amplitude of a signal component corresponding to a period subsequent to the period Td (i.e., a period corresponding to the maximum amplitude and later periods) decreases over time. Since TSPs and TSP response signals (SR 1  to SRk) are repeated for every unit period TN, it is possible to consider that the signal component corresponding to the periods Td follows the trailing end of the waveform in  FIG. 11 . Therefore, the amplitude in the period Td can also be considered as decreasing over time.  
      By utilizing the above-described properties of reverberation characteristic ((A)to (C)), the threshold level V TH  for detecting a rising edge of an impulse response can be determined in accordance with an algorithm described below.  
      Specifically, since the signal component in the period Td can be considered as subsequent to the trailing end of the waveform in  FIG. 11 , the period Td and a predetermined period Tt in the trailing part of the waveform are set as a detection period Tx for detecting the level of the background noise. The period Tt serves to provide a sufficient detection period in a case where the delay period Td is short.  
      Referring to  FIG. 12 , an example of a characteristic diagram for determining the threshold level V TH  is shown. The abscissa represents the maximum amplitude value Dx_max of an inverse-TSP impulse response signal in the detection period Tx, and the ordinate represents the threshold level V TH . A maximum value SR_max in the ordinate represents the maximum amplitude of the impulse response signal which corresponds to the rising edge.  
      In this characteristic diagram, the following conditions are satisfied:  
      (D) in section A (Dx_max≦SR_max·2.5%),  
      V TH =SR_max·5%  
      (E) in section B (SR_max·2.5%&lt;Dx_max≦SR_max·5%),  
      SR_max·5%&lt;V TH ≦SR_max·20%  
      (F) in section C (SR_max·5%&lt;Dx_max≦SR_max·7.5%),  
      SR_max·20%&lt;V TH ≦SR_max·80%  
      (G) in section D (SR_max·7.5%&lt;Dx_max)  
      V TH =SR_max·80%.  
      In the section A in the abscissa, the threshold level V TH  is set by a fixed ratio (=5%) and is not associated with the maximum amplitude value Dx_max. Such a fixed ratio is used since noise is potentially present in a reproduction sound field, and the noise level of the sound field can be assumed to not exceed a predetermined level from a statistical point of view. In the section D, the threshold level V TH  is set to be 80% which is close to the maximum value. Two-phase gradients corresponding to sections B and C serve for transitioning between the section A and the section D.  
      As described above, the threshold level V TH  is dynamically changed in accordance with the noise level in the detection period Tx. This reduces possibility that an amplitude change which occurs prior to the actual rise time of an impulse response is misrecognized as corresponding to the rising edge of the impulse response.  
      [4] System Configuration  
       FIG. 13  illustrates a sound field correcting apparatus to which an embodiment of the present invention is applied. This sound field correcting apparatus is implemented as an adapter type for a known multi-channel AV (Audio/Visual) reproducing apparatus.  
      [4-1] AV Reproducing Apparatus  
      In  FIG. 13 , the AV reproducing apparatus includes a signal source  11  for generating an AV signal, a display  12 , a digital amplifier  13 , and speakers  14 C to  14  RB. The signal source  11  may be a DVD player, a tuner for satellite broadcasting, or the like. The signal source  11  has a DVI (Digital Visual Interface) output, and a video signal DV is output as a digital signal. At the same time, digital audio signals for seven channels are encoded into a serial signal DA for output.  
      The display  12  has a DVI input. Thus, normally, the digital video signal DV output from the sound source  11  can be directly input to the display  12 . The digital amplifier  13  includes a multi-channel decoder and is configured as a so-called class D amplifier. Specifically, it is normally possible to input the digital audio signal DA output from the sound source  11  to the digital amplifier  13 . In addition, the digital amplifier  13  separates (decodes) the signal DA into signals for the individual channels and performs class D power amplification on the channel signals so as to output analog audio signals for the individual channels.  
      The audio signals output from the amplifier  13  are supplied to individual speakers  14 C to  14 RB corresponding to the channels. The speakers  14 C to  14 RB are installed positions in the center front, left front, right front, left side, right side, left back, and right back, respectively, with respect to a listener.  
      [4-2] Sound Field Correcting Apparatus  
      [4-2-1] Configuration of Sound Field Correcting Apparatus  
      In  FIG. 13 , a sound field correcting apparatus  20  according to an embodiment of the present invention is illustrated. The sound field correcting apparatus  20  is connected to a signal line between the signal source  11 , and the display  12  and the digital amplifier  13 . The digital video signal DV output from the signal source  11  is supplied to the display  12  through a delay circuit  21 . The delay circuit  21  includes a field memory or the like and provides the video signal DV a delay of a period based on a delay of the digital audio signal DA due to sound field correction, so as to synchronize an image and reproduced sound (i.e., lip-sync).  
      In addition, in the sound field correcting apparatus  20 , the digital audio signal DA is supplied to a decoder  22  and separated into audio signals DC to DRB for the individual channels. The audio signal DC for a center channel is supplied to a correction circuit  23 C. This correction circuit  23 C includes an equalizer circuit  231  and a switch circuit  232 . The audio signal DC from the decoder  22  is supplied to the switch circuit  232  through the equalizer circuit  231 .  
      In this case, the equalizer circuit  231  is constituted by, for example a DSP (Digital Signal Processor) and controls the delay characteristic, frequency characteristic, a phase characteristic, level, etc., of the audio signal DC, so as to perform sound field correction on the signal DC. The switch circuit  232  has such connection depicted in the figure during a normal audio/visual operation. In measurement and analysis operations for sound field correction, the switch circuit  232  has a connection state which is inverted to that depicted in the figure. Thus, in a normal audio/visual operation, the audio signal DC, which has undergone sound field correction, is supplied from the equalizer circuit  231  and then output from the switch circuit  232 . The audio signal DC is then fed to an encoder  24 .  
      The remaining audio signals DL to DRB, which has been separated by the decoder  22 , are fed to the encoder  24  through correction circuits  23 L to  23 RB, respectively. Each of the correction circuits  23 L to  23 RB has the same configuration as the correction circuit  23 C. Thus, in a normal audio/visual operation, the audio signals, which have undergone sound field correction, are output from the correction circuits  23 L to  23 RB, respectively, and then supplied to the encoder  24 .  
      Then, in the encoder  24 , the audio signals DC to DRB for the individual channels are converted into a serial signal DS, and this serial signal DS is supplied to the digital amplifier  13 . Thus, in a normal audio/visual operation, the audio signal DA output from the signal source  11  undergoes sound field correction through the correction circuits  23 C to  23 RB and then is supplied to speakers  14 C to  14 RB. As a result, the audio signal DA is emitted from the speakers  14 C to  14 RB as reproduction sound which has been corrected so as to be suitable to an environment where the speakers are arranged.  
      The sound field correcting apparatus  20  also includes a TSP signal forming circuit  31 . The TSP signal forming circuit  31  includes a memory to which a TSP signal is written in the form of digital data and a readout circuit for reading the digital data. The TSP signal forming circuit  31  outputs a TSP signal repeatedly for each unit period over the periods T 1  to Tk, in accordance with control performed by a controller  35 . The TSP signal is supplied to the switch circuits  232  of the correction circuits  23 C to  23 RB.  
      In a measurement of a sound field in an acoustic state, a microphone  15  is provided at the listener&#39;s position so that TSP sound is picked up. At this time, the microphone  15  is arranged so that its diaphragm is in a horizontal plane so as to be nondirectional. Thus, the microphone  15  has a constant sensitivity regardless of the position and orientation of the speakers.  
      An output signal SR 1  of the microphone  15  is supplied to an analog/digital (A/D) converter  33  through a microphone amplifier  32  and then converted into a digital signal SRi with a sampling frequency of 48 kHz, for example. This digital signal SR 1  is supplied to an analysis circuit  34 .  
      The analysis circuit  34  includes a memory  341  and a DSP  342 . When the emission of TSP sound is initiated, the DSP  342 , using the memory  341 , accumulates and averages the output signals SR 1  for every unit period TN (for example, 4096-sample period) over the period T 1  to T (k+L). Thus, in the end of the period T(k+L) a wrapped signal Sw is provided to the memory  341 .  
      The wrapped signal Sw is analyzed through a scheme described in the foregoing ([1-2]) by the DSP  342 , and the result of the analysis is supplied to the controller  35 . The controller  35  has a microcomputer so as to perform control of formation of TSP signals in the TSP signal forming circuit  31  and switching of the switch circuits  232 . The controller  35  also performs setting of the equalizer circuits  231  of the correction circuit  23 C to  23 RB in accordance with the analysis result obtained from the analysis circuit  34 .  
      The controller  35  is connected to various operation switches  36  as user interfaces and to a display device such as an LCD panel  37  for displaying an analysis result or the like.  
      [4-2-2] Operation in Analysis Processing of Sound Field Correcting Apparatus  20   
      When a setting switch, which is one of the operation switches  36 , is operated, the controller  35  inverts the connection of the switch circuits  232  of the correction circuits  23 C to  23 RB. The controller  35  also controls the TSP signal forming circuit  31 , so that a TSP signal is supplied to the switch circuit  232  of the correction circuit  23 C. Thus, TSP sound is output over the period T 1  to Tk from the speaker  14 C. At this time, no sound is output from the speakers of the other channel.  
      At this time, the TSP sound emitted from the speaker  14 C is picked up by the microphone  15 . The controller  35  controls the analysis circuit  34  so that analysis processing is initiated. Through this analysis processing, parameters such as the distance between the speaker  14 C to the microphone  15  and the frequency characteristic are calculated, and the result is provided to the controller  35 . On the basis of the result of the analysis processing, the controller  35  sets the equalizer circuit  231  for sound field correction. Then, the switch circuit  232  are set in the state depicted in the figure, and thus the sound field correction processing for the signal DC for the corresponding channel is terminated. Likewise, setting of sound field correction for the other channels are performed.  
      Consequently, in a normal audio/visual operation, sound field correction is performed on the audio signals DA output from the signal source  11  by the correction circuits  23 C to  23 RB. Then, the corrected signals are supplied to the speakers  14 C to  14 RB, and thus the speakers  14 C to  14 RB output reproduced sound which has been corrected so as to be suitable to the environment where the speakers are arranged.  
      [5] Other Implementations  
      The values used for define the sections A to D (i.e., 0.025, 0.05, and 0.075) and the values used for sectioning the threshold level V TH  (i.e., 5%, 20%, and 80%), which are described with reference to  FIG. 12 , can be other than the those illustrated. In addition, the maximum amplitude values Dx_max may be obtained by squaring an instantaneous value in the detection period Tx or the absolute value of the instantaneous value. Further, although the characteristic in  FIG. 12  is indicated by broken lines, a characteristic function indicated by a curve can also be employed. Thus, any characteristic can be employed as long as it serves to determine the threshold level on the basis of data such as a maximum value and average energy in the detection period Tx.  
      Moreover, to increase accuracy in actual implementation, the threshold level V TH  can be configured to be two-phase. For example, a high threshold level V THH  is set as a reference threshold level. Then, with the above-described technique, level determination is performed forward along the time axis so that a rising edge is obtained as a dummy rising edge. Subsequently, from the dummy rising edge, level determination is performed backward along the time axis, and a time point where the threshold level is lower than a threshold level T HL  (V THL &lt;V THH ) is determined to be the actual rising edge. Alternatively, it may also be possible that the level determination is performed backward along the time axis from the dummy rising edge for a predetermined sample value, and a time point that gives the closest value to the dummy rising edge is determined to be the actual rising edge.  
      Further, it is also possible that the analysis processing is performed on a wrapped signal Sw or an impulse response signal obtained by inversely TSP-transforming the wrapped signal Sw, after is filtered so that the effect of noise and excessive fluctuation of waveform is reduced.  
      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.