Patent Publication Number: US-2017353169-A1

Title: Signal processing apparatus and signal processing method

Description:
CROSS REFERENCE 
     This Nonprovisional application claims priority under 35 U.S.C. §119(a) on Patent Application No. 2016-109951 filed in Japan on Jun. 1, 2016, the entire contents of which are hereby incorporated by reference. 
     BACKGROUND OF THE INVENTION 
     1. Field of the Invention 
     Some preferred embodiments of the present invention relate to a signal processing apparatus and a signal processing method that are capable of calculating a gain correction amount by analyzing an input signal. 
     2. Description of the Related Art 
     In facilities such as a concert hall, music of various genres may be performed or a speech such as a lecture may be delivered. Such facilities require various acoustic characteristics (reverberation characteristics, for example). For example, a performance requires comparatively long reverberation while a speech requires comparatively short reverberation. 
     However, in order to physically change reverberation characteristics in a concert hall, the size of an acoustic space needs to be changed by moving a ceiling, for example, so that very large-scale equipment has been necessary. 
     Accordingly, a sound field control device disclosed in Japanese Unexamined Patent Application Publication No. H06-284493, for example, performs processing to support a sound field by processing a sound collected by a microphone through an FIR filter to generate a reverberant sound and outputting the reverberant sound from a speaker installed in a concert hall. 
     However, in such a sound field control device, the sound that has been output from the speaker is collected again by the microphone through the transmission system of an acoustic space and processed by the FIR filter and then is output from the speaker. In other words, the sound field control device includes an acoustic feedback system. Therefore, if acoustic field support is performed, a specific frequency component may increase and howling or coloration may occur. 
     Accordingly, an acoustic field support device disclosed in Japanese Unexamined Patent Application Publication No. 2012-060333, for example, performs processing to reduce howling or coloration by performing signal processing in which the amplitude characteristics of an impulse response are smoothed. 
     However, the cause of coloration is not only due to the transmission system of an acoustic space. Coloration includes: coloration that occurs because, due to original standing waves in the acoustic space, specific standing waves remain for a long time in an attenuation process when an impulse is generated in a room; and coloration due to the acoustic feedback of a system. Examples of coloration due to the acoustic feedback of a system includes a case in which, when a sudden sound occurs, the sudden sound is amplified by signal processing and a specific frequency component may increase. 
     SUMMARY OF THE INVENTION 
     In view of the foregoing, preferred embodiments of the present invention are directed to provide a signal processing apparatus and a signal processing method that are capable of reducing both coloration due to the transmission system of an acoustic space and coloration due to signal processing. 
     A signal processing apparatus according to a preferred embodiment of the present invention includes: an obtaining portion configured to obtain an impulse response in a semi-open state in which, among a plurality of acoustic feedback systems, at least one acoustic feedback system is open and at least one acoustic feedback system is closed; and a calculating portion configured to calculate a gain correction amount based on the impulse response that the obtaining portion has obtained and to output a calculated gain correction amount. 
     The signal processing apparatus is configured to reduce both coloration due to the transmission system of an acoustic space and coloration due to signal processing. 
     The above and other elements, features, characteristics, and advantages of the present invention will become more apparent from the following detailed description of the preferred embodiments with reference to the attached drawings. 
    
    
     
       BRIEF DESCRIPTION OF THE DRAWINGS 
         FIG. 1  is a schematic transparent perspective view of an acoustic space. 
         FIG. 2  is a block diagram illustrating a configuration of a sound field support (AFC: Active Field Control) system. 
         FIG. 3  is a block diagram illustrating the AFC system and a Personal Computer (PC). 
         FIG. 4  is a flow chart showing an operation of a signal processing apparatus. 
         FIG. 5  is a block diagram illustrating a configuration of the AFC system in an open state. 
         FIG. 6  is a block diagram illustrating a configuration of the AFC system in a semi-open state. 
         FIG. 7A  and  FIG. 7B  illustrate an impulse response according to a frequency. 
         FIG. 8A ,  FIG. 8B , and  FIG. 8C  illustrate amplitude correction processing. 
         FIG. 9A  and  FIG. 9B  are graphs showing frequency characteristics. 
         FIG. 10  illustrates a comparison between the presence and absence of correction processing. 
         FIG. 11  is a graph showing a result of quantitative evaluation. 
         FIG. 12  is a flow chart showing another operation of the signal processing apparatus. 
     
    
    
     DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS 
     A signal processing apparatus according to a preferred embodiment of the present invention includes: an obtaining portion configured to obtain an impulse response in a semi-open state in which, among a plurality of acoustic feedback systems, at least one acoustic feedback system is open and at least one acoustic feedback system is closed; and a calculating portion configured to calculate a gain correction amount based on the impulse response that the obtaining portion has obtained and to output a calculated gain correction amount. 
     Thus, the signal processing apparatus, in order to obtain an impulse response in a semi-open state, may obtain transmission characteristics including an acoustic space and signal processing. Therefore, the signal processing apparatus is able to reduce not only coloration due to the transmission system of an acoustic space but also coloration due to signal processing. 
       FIG. 1  is a schematic transparent perspective view of an acoustic space.  FIG. 2  is a block diagram illustrating a configuration of a sound field support (AFC: Active Field Control) system. 
     In the acoustic space, a microphone  11 A, a microphone  11 B, a microphone  11 C, a microphone  11 D, a speaker  51 A, a speaker  51 B, a speaker  51 C, a speaker  51 D, a speaker  51 E, and a speaker  51 F are installed. 
     While, in this example, four microphones are installed, the AFC system  1  is able to operate as long as at least one or more microphones are installed. Similarly, the number of speakers is not limited to six, either, and, as long as at least one or more speakers are installed, the AFC system  1  is able to operate. 
     The microphone  11 A, the microphone  11 B, the microphone  11 C, and the microphone  11 D are installed on a ceiling immediately above a sound source  61 . The microphone  11 A, the microphone  11 B, the microphone  11 C, and the microphone  11 D mainly collect sound that the sound source  61  emits. 
     The speaker  51 A, the speaker  51 B, the speaker  51 C, the speaker  51 D, the speaker  51 E, and the speaker  51 F are installed in the vicinity of the ceiling immediately above a listener  65 . It is to be noted that the installation positions of the microphones and the speakers are not limited to this example. 
     As illustrated in  FIG. 2 , the AFC system  1  is provided with a front end circuit (HA&amp;AD)  21 , a microphone assigning portion (MIC Assign)  22 , an FIR filter  23 , an equalizer (EQ)  24 , a level matrix (Level Matrix)  25 , an EQ  26 , a DA converter  27 , a power amplifier (Power Amp)  28 , a controller  30 , and a storage portion  31 . 
     The front end circuit  21  contains a microphone amplifier and an AD converter. The front end circuit  21  amplifies an analog signal that the microphone  11 A, the microphone  11 B, the microphone  11 C, and the microphone  11 D have output and outputs the analog signal as a digital signal. 
     The microphone assigning portion  22  has the function of an EMR (Electronic Microphone Rotator). The EMR is a function to switch with a lapse of time a connection relationship between digital signals of four channels to be input and digital signals of four channels to be output. Accordingly, the microphone assigning portion  22  flattens frequency characteristics of an acoustic feedback system from an acoustic space  62  to the acoustic space  62  back again through a microphone, signal processing, amplification processing, and a speaker. 
     The FIR filter  23  convolves an impulse response to the digital signals of the four channels to be input and generates a reverberant sound. The storage portion  31  stores data related to the impulse response. The controller  30  reads the data related to a predetermined impulse response from the storage portion  31 , and sets a filter coefficient corresponding to the impulse response to the FIR filter  23 . 
     The EQ  24  includes a plurality of parametric equalizers (PEQ), for example. The EQ  24  corrects the gain of a predetermined bandwidth (Q value) around a specified frequency of each of the digital signals of the four channels to be input. The controller  30  specifies a center frequency, a Q value, and a gain. 
     The level matrix  25  distributes the digital signals of the four channels to be input, to six output channels. The level matrix  25  also performs a gain adjustment and a delay adjustment of each of the output channels. The controller  30  specifies a gain and delay of each of the output channels. 
     The EQ  26  corrects the frequency characteristics of each of the digital signals of the six channels to be input from the level matrix  25 . 
     The DA converter  27  converts each of the digital signals of the six channels to be output from the EQ  26 , to an analog signal. 
     The power amplifier  28  amplifies each analog signal that has been output from the DA converter  27 , and outputs amplified analog signals to the speaker  51 A, the speaker  51 B, the speaker  51 C, the speaker  51 D, the speaker  51 E, and the speaker  51 F, respectively. 
     The controller  30  reads a program stored in the storage portion  31  and collectively controls the AFC system  1 . In the present preferred embodiment, the storage portion  31  may be configured by a volatile memory, a nonvolatile memory, an HDD, an SSD, or the like. The controller  30  implements the functions of the obtaining portion  151  and the calculating portion  152  by causing the CPU  301  to execute the program. In other words, the controller  30  is equivalent to the signal processing apparatus of the present invention, and the CPU  301  is equivalent to the obtaining portion  151  and the calculating portion  152 . 
     It is to be noted that the function of the controller  30 , as illustrated in  FIG. 3 , is also able to be implemented by an external device (a PC  100  in the present preferred embodiment). In the example of  FIG. 3 , the PC  100  connected to the AFC system  1  is provided with a controller  101 . The controller  101  is implemented when the CPU  105  of the PC  100  executes an application program. The controller  101  controls the various configurations of the AFC system  1 . In addition, in this example, the obtaining portion  151  and the calculating portion  152  are implemented as the function of the application program (a tuning tool  102 ) that the CPU  105  of the PC  100  executes. The tuning tool  102  is equivalent to the signal processing apparatus of the present invention. 
     The obtaining portion  151  obtains an impulse response to be described later, and the calculating portion  152 , based on an obtained impulse response, calculates a parameter (gain correction amount) of the EQ  24  and outputs the parameter to the EQ  24 . 
       FIG. 4  is a flow chart showing an operation of the AFC system  1 . To begin with, the AFC system  1  performs an automatic adjustment (a coarse adjustment) (s 11 ). The coarse adjustment is to perform processing of measuring the impulse response of the acoustic space  62 , detecting a frequency at which howling may occur, and reducing the gain of the frequency. 
     In such a case, as illustrated in  FIG. 5 , the controller  30  controls the microphone assigning portion  22 , stops the output of a signal, and makes an open state. It is to be noted that, while this example illustrates a mode in which the controller  30  controls the microphone assigning portion  22  to make an open state, it is also possible to employ a mode in which the output of a signal in any one of blocks is stopped to make an open state. Moreover, it is also possible to make an open state by installing a switch between any blocks up to the level matrix  25  and turning off the switch. 
     Then, the controller  30  outputs a measurement sound (impulse sound) to one of the four channels, inputs the measurement sound through a microphone, and obtains an impulse response. The controller  30  converts an obtained impulse response into a frequency signal by a method such as the FFT. The controller  30  detects a frequency of a peak that indicates a remarkably high level on a frequency axis. The controller  30  may detect a frequency that indicates a level equal to or above a predetermined threshold value, for example, as a peak frequency. At the end, the controller  30  sets a center frequency, a Q value, and a gain to the EQ  24  so as to reduce the level of a detected peak frequency. 
     The controller  30  performs a coarse adjustment by performing the above measurement with respect to all four input channels. Accordingly, the controller  30  reduces howling from occurring, and stabilizes the state of the AFC system  1 . 
     Subsequently, returning to  FIG. 4 , the controller  30  obtains the impulse response of each channel in a semi-open state (s 12 ). 
     In such a case, as illustrated in  FIG. 6 , the controller  30  controls the microphone assigning portion  22 , opens one channel to be measured, and closes the other channels. It is to be noted that, as described above, the controller  30  may control the microphone assigning portion  22  or the controller  30  may stop the output of a signal in any of the blocks to make a semi-open state. In addition, it is also possible to make a semi-open state by installing a switch between any blocks up to the level matrix  25  and turning on and off the switch. 
     At this time, the function of the EMR in the microphone assigning portion  22  stops. In other words, the digital signals that have been input from the microphone of each of the channels are respectively output directly in the channels. However, such processing may be performed while the function of the EMR is kept executed. 
     The controller  30  outputs a measurement sound (impulse sound) to the channel that has been made open, inputs the measurement sound through a microphone, and obtains an impulse response. Thus, the processing of obtaining an impulse response in a semi-open state may be executed by the function of the obtaining portion  151  in the controller  30 . On the other hand, the processing at step s 13  and the following steps shown in the flow chart of  FIG. 4  may be executed by the function of the calculating portion  152  in the controller  30 . 
     Subsequently, the controller  30 , with respect to an obtained impulse response, may cut a band below 200 Hz, and may extract a range of 200 Hz and above (s 13 ). 
       FIG. 7A  and  FIG. 7B  illustrate an impulse response according to a frequency. The vertical axis represents a frequency and the horizontal axis represents time. 
     As illustrated in  FIG. 7A , in the low frequency band, since a level due to background noise in the acoustic space  62  is large, it is difficult to analyze the influence of coloration. Therefore, the controller  30  may perform processing of cutting the band below 200 Hz and extracting the band of 200 Hz and above in which the influence of coloration is large. 
     In addition, the controller  30  extracts a predetermined level range (−30 dB to −50 dB, for example) in a reverberation attenuation waveform (see  FIG. 8B ) calculated from the obtained impulse response (s 14 ). For example, in the example of  FIG. 7B , a range from 1.2 sec. to 2.2 sec. corresponds to the range from −30 dB to −50 dB. As illustrated in  FIG. 7B , since a high level signal is input for a while after sound is input directly, it is difficult to analyze the influence of coloration. In contrast, when a low level signal is input, the influence of background noise becomes larger. Accordingly, the controller  30 , by extracting a time zone in which the level of the obtained impulse response is a predetermined level range (−30 dB to −50 dB, for example), may define a time zone in which the influence of coloration is large, as a processing target. 
     Subsequently, the controller  30  performs non-linear attenuation correction to an extracted impulse response (s 15 ). The non-linear attenuation correction is to perform processing of raising a gain with a lapse of time so that the level of an impulse response does not attenuate (see Hanyu et al., “Calculation of Attenuation Removed Impulse Response of Indoor Sound Field with Non-Linear Attenuation,” The Acoustical Society of Japan, lecture paper, March 2014). 
       FIG. 8A  illustrates an impulse response (linear scale) that has been cut in a target level range, and  FIG. 8B  illustrates a reverberation attenuation curve calculated from the impulse response (logarithmic scale). As illustrated in  FIG. 8A , the level of the impulse response, on a linear scale, decreases exponentially with a lapse of time. In addition, as illustrated in  FIG. 8B , the impulse response, also on a logarithmic scale, changes not with linear attenuation but with a lapse of time. 
     Accordingly, the controller  30  performs a level adjustment according to attenuation characteristics of the impulse response so that the level of the impulse response may not attenuate. In other words, the controller  30  sets a gain of the inverse characteristics to the attenuation characteristics of the impulse response. In particular, the controller  30  may preferably calculate the attenuation characteristics of the impulse response in each case in a finely divided time range (in a range of 0.5 sec., for example) and obtain a level correction value. For example, at each time in the above mentioned Hanyu method, a short-time attenuation factor is calculated in a section of ±5 dB. 
     Accordingly, as illustrated in  FIG. 8C , the impulse response may be an attenuation removed IR of which the level does not attenuate with a lapse of time. 
     Subsequently, the controller  30  converts a calculated attenuation removed IR into a frequency signal by a method such as the FFT (s 16 ).  FIG. 9A  is a graph showing the frequency characteristics (linear scale) of an attenuation removed IR, and the characteristics of a moving average, and  FIG. 9B  is a graph showing the frequency characteristics (logarithmic scale) of an attenuation removed IR, and the characteristics of a moving average. 
     The controller  30 , from the frequency characteristics of the attenuation removed IR as shown in  FIG. 9A  and  FIG. 9B , calculates a target frequency that influences coloration. 
     The controller  30  may first calculate a moving average, for example, with respect to the frequency characteristics after performing the FFT in the processing of step s 16  (s 17 ). The controller  30  calculates an average value of amplitude, for example, while moving a frequency band in the one-third octave width. Any method may be used as long as it can smooth the frequency characteristic of the attenuation removed IR, not limited to the moving average. In addition, the controller  30  extracts a predetermined number (eight in the present preferred embodiment) of peaks sequentially from a peak with the highest amplitude value (s 18 ). Then, the controller  30 , with respect to each of the extracted eight peaks, calculates a difference between an amplitude value and a value of the moving average (s 19 ). Subsequently, the controller  30  rearranges the extracted eight peaks in descending order of amplitude (s 20 ). It is to be noted that, the controller  30 , in a case in which, sequentially from a peak with the highest amplitude value, a peak of which the level is relatively high is set as a standard and other peaks are in a predetermined band (the one-third octave width, for example) around the frequency of the peak as the standard, may perform processing of excluding the other peaks (s 21 ). 
     At the end, the controller  30 , with respect to the frequency of the peak that has remained after the processing of step s 21 , obtains a difference between an amplitude value and the above moving average, calculates a gain correction amount, and applies the gain correction amount to a corresponding channel in the EQ  24  (s 22 ). The gain correction amount is set to a value such that the amplitude value of each peak is a moving average value +10 dB, for example. It is to be noted that, while a Q value is arbitrary, the controller  30  sets the greatest Q value that the EQ  24  is able to set, in the present preferred embodiment of the present invention. 
     With the above processing, in the EQ  24 , the gain of the frequency that influences coloration is reduced, so that coloration is able to be reduced. In particular, the controller  30 , in order to obtain an impulse response in a semi-open state, performs coloration suppression processing including the signal processing of the AFC system  1 . 
     The controller  30  performs the above processing with respect to each of the four channels and sets a gain correction amount of each of the channels in the EQ  24 . It is to be noted that, at the time of actual operation, the EMR functions in the microphone assigning portion  22 . Therefore, the controller  30 , with respect to all connection configurations by the EMR, may preferably calculate a target frequency of coloration and may preferably calculate a gain correction amount. In addition, as described above, the controller  30 , in a state in which the function of the EMR in the microphone assigning portion  22  is executed, may obtain an impulse response. 
     Moreover, while, in the above example, a mode in a semi-open state in which a channel to be analyzed is open and all the other channels are closed is described, the controller  30  may perform various types of processing in a semi-open state in which at least one acoustic feedback system (channel to be analyzed) is open and at least one acoustic feedback system is closed. 
     Subsequently,  FIG. 10  illustrates a comparison between the presence and absence of correction processing (illustrates an impulse response according to a frequency). As illustrated in  FIG. 10 , the impulse response before correction has a portion that does not attenuate in some frequencies even after several seconds pass. The frequency components that do not attenuate may be coloration. However, in the impulse response after correction, the frequency components are reduced. Thus, coloration is reduced by correction processing. 
     In addition,  FIG. 11  is a graph showing a result of the quantitative evaluation of coloration. The horizontal axis of the graph represents the standard deviation of frequency characteristics. As a frequency component that represents a high level peak increases, the standard deviation becomes larger. The vertical axis of the graph represents a psychological scale and corresponds to the percentage of people who feel coloration. 
     As shown in  FIG. 11 , the standard deviation has a high correlation with the occurrence of coloration. In other words, the graph of  FIG. 11  shows that coloration is reduced as the standard deviation decreases. The “OFF” in  FIG. 11  indicates a state in which the AFC system  1  is not in operation. In a case in which the AFC system  1  is not in operation, only a natural reverberant sound in the acoustic space  62  occurs without an acoustic feedback system, so that people who feel coloration are extremely small. On the other hand, the “ON: WITHOUT CORRECTION” on the right side of  FIG. 11  indicates a state in which the AFC system  1  is operated after the coarse adjustment shown at step s 11  of  FIG. 4  is performed. When the AFC system  1  is operated, an acoustic feedback system is formed, so that the standard deviation becomes larger and the percentage of people who feel coloration increases. However, as shown in the “ON: WITH CORRECTION” in  FIG. 11 , if the various types of processing at step s 12  and the following steps shown in the present preferred embodiment are performed, the standard deviation becomes smaller and the percentage of people who feel coloration decreases. 
     Therefore, the controller  30 , by obtaining an impulse response in a semi-open state and calculating a gain correction amount based on an obtained impulse response, is able to reduce both coloration due to the transmission system of an acoustic space and coloration due to signal processing. 
     Subsequently,  FIG. 12  is a flow chart showing an operation of the controller  30  according to a modification. Like reference numerals are used to indicate processing common to the processing shown in  FIG. 4 , and the associated description will be appropriately omitted. 
     In the modification, the controller  30  rearranges extracted eight peaks, in descending order of the difference calculated at step s 19  (s 30 ). In other words, in the modification, priority is given not to the size of the amplitude value of each peak but to a large difference with a moving average. Accordingly, the controller  30  is able to perform more natural correction by hearing. 
     It is to be noted that, while an example in which a high level peak frequency is set to be a target frequency of coloration is shown in the present preferred embodiment, a method of extracting the target frequency of coloration is not limited to this example. For example, with respect to an obtained impulse response, power for each predetermined frequency band (one-third octave width, for example) is measured, and, when measured power exceeds a predetermined threshold value, processing of reducing the frequency band may be performed. In addition, a frequency of which the difference with a moving average is smaller than a predetermined value may be specified and another frequency other than the frequency may be set as a target frequency of coloration. 
     Finally, the foregoing preferred embodiments are illustrative in all points and should not be construed to limit the present invention. The scope of the present invention is defined not by the foregoing preferred embodiment but by the following claims. Further, the scope of the present invention is intended to include all modifications within the scopes of the claims and within the meanings and scopes of equivalents.