Abstract:
A howling canceler apparatus is used in a sound amplification system having a sound amplifier which connects with a multiple of speakers and one or more of microphones. In the howling canceler apparatus, a plurality of adaptive filters are provided in correspondence to a plurality of feedback transmission paths which are formed between each of the multiple of the speakers and each of the one or more of the microphones. Each adaptive filter is set with a filter coefficient simulating a transfer function of the corresponding feedback transmission path for processing the output sound signal to generate a simulation signal simulating a feedback sound traveling through the corresponding feedback transmission path. Each adaptive filter is capable of setting its own filter coefficient based on the output sound signal and a residual signal. A subtraction portion subtracts the simulation signal outputted from the adaptive filter from the input sound signal inputted from the microphone to generate the residual signal, and outputs this residual signal to the adaptive filter and to the sound amplifier.

Description:
BACKGROUND OF THE INVENTION  
       [0001]     1. Technical Field  
         [0002]     The present invention relates to a howling canceler apparatus to prevent howling which is caused by supplying a microphone with feedback sounds from multiple speakers, and also relates to a sound amplification system equipped with this howling canceler apparatus.  
         [0003]     2. Related Art  
         [0004]     A sound amplification system amplifies a sound signal input from a microphone and inputs the amplified sound signal to a speaker. It is widely known that the sound amplification system forms a closed loop along a path from the speaker to the microphone and howling is generated by repeatedly amplifying a feedback sound signal that is output from the speaker and is input to the microphone.  
         [0005]     To prevent such howling, it has been proposed that an adaptive filter is used to generate a simulation signal simulating a feedback sound signal, and the sound amplification system uses a howling canceler apparatus having such an adaptive filter to subtract the simulation signal from an input signal supplied from the microphone (See Inazumi, Imai, and Konishi, “Prevention of acoustic feedback in the sound amplification system using the LMS algorithm,” lecture thesis collection pp. 417-418, The Acoustical Society of Japan, March, 1991). Constituent portions of the howling canceler apparatus operate as follows.  
         [0006]     When a sound signal is input to a speaker, the same sound signal as that sound signal is input to a delay portion. The delay portion delays the sound signal for a delay time spent by the sound signal traveling from the speaker to a microphone. A convolution operation is performed for the delayed signal using a filter coefficient of the adaptive filter to generate a simulation signal. A subtraction portion subtracts the simulation signal from the signal input from the microphone to leave a residual signal that is then output to a sound amplification portion. The sound amplification portion amplifies the residual signal that is then input to the speaker. The speaker generates sound. The adaptive filter is supplied with the residual signal as a reference signal. A known adaptive algorithm (e.g., LMS (Least Mean Square) algorithm) is used to update the filter coefficient (filter characteristic) so that the residual signal is minimized. In this manner, the adaptive filter&#39;s filter coefficient approximates to a transfer function of the feedback transmission path from the speaker to the microphone. The filter coefficient is used to simulate the feedback transmission path&#39;s transfer function. The signal processed by the adaptive filter, i.e., the simulation signal approximates to a feedback sound signal. This makes it possible to remove feedback sound signal components from the input sound signal and prevent the howling.  
         [0007]     When multiple speakers are connected, however, a conventional sound amplification system may not be able to stably (statically determinately) simulate transfer functions using the adaptive filter. In this configuration, the sound output from multiple speakers may be input to the same microphone. The same microphone is supplied with feedback sounds transferred by multiple feedback transmission paths. When the same adaptive filter is used to simulate transfer functions for the multiple feedback transmission paths, the transfer functions cannot be simulated stably, making it difficult to accurately prevent the howling.  
       SUMMARY OF THE INVENTION  
       [0008]     It is therefore an object of the present invention to provide a howling canceler apparatus and a sound amplification system capable of stably simulating transfer functions using adaptive filters and accurately preventing howling even in an acoustic system configuration where multiple feedback paths are formed from speakers to microphones.  
         [0009]     To solve the above-mentioned problem, the present invention incorporates the following means.  
         [0010]     (1) The present invention provides a howling canceler apparatus included in or connected with a sound amplification system having a sound amplification portion which connects with a multiple of speakers and one or more of microphones and which amplifies an input sound signal inputted from the microphone and supplies the amplified sound signal as an output sound signal to the speakers. The howling canceler apparatus comprises: a plurality of adaptive filters which are provided in correspondence to a plurality of feedback transmission paths which are formed between each of the multiple of the speakers and each of the one or more of the microphones, each adaptive filter being set with a filter coefficient simulating a transfer function of the corresponding feedback transmission path for processing the output sound signal to generate a simulation signal simulating a feedback sound traveling through the corresponding feedback transmission path, each adaptive filter being capable of setting its own filter coefficient based on the output sound signal and a residual signal; and a subtraction portion which subtracts the simulation signal outputted from the adaptive filter from the input sound signal inputted from the microphone to generate the residual signal, and which outputs this residual signal to the adaptive filter and to the sound amplification portion as the input sound signal.  
         [0011]     According to the embodiment, the sound amplification system is connected with multiple speakers and one or more microphones. There may be multiple feedback transmission paths between the speakers and the microphones as many as combinations of the speakers and the microphones. That is, there may be feedback transmission paths between the speakers and the microphones for “the number of speakers multiplied by the number of microphones”.  
         [0012]     According to the configuration of the present invention, the howling canceler apparatus has the adaptive filter for each of the multiple feedback transmission paths. The adaptive filter sets a filter coefficient based on the output sound signal and the residual signal. The filter coefficient simulates the transfer function for the corresponding feedback transmission path. The adaptive filter is supplied with an output sound signal to be output to the speaker. The adaptive filter processes the output sound signal to generate a simulation signal that simulates the signal associated with the feedback sound supplied from the feedback transmission path. Even when the microphone is supplied with input sound signals via multiple feedback transmission paths, each adaptive filter only needs to simulate the transfer function for one feedback transmission path. This makes it possible to stably simulate the transfer function for the feedback transmission path in comparison with the conventional technology that simulates multiple feedback transmission paths using a single or common adaptive filter.  
         [0013]     The subtraction portion subtracts the simulation signal output from the adaptive filter from the input sound signal supplied from the microphone to generate a residual signal. This residual signal is output to the adaptive filter and to the sound amplification portion as the input sound signal. The sound amplification portion can amplify the input sound signal while feedback sound components are fully removed. Accordingly, it is possible to effectively prevent the howling from occurring due to repeated amplification of feedback sound components.  
         [0014]     (2) According to the present invention, the above-mentioned howling canceler apparatus is provided with a correlation reduction process portion which decreases correlation among a multiple of the output sound signals, and then feeds these output sound signals after the correlation is decreased to the speakers and the adaptive filters. For example, let us suppose that the speakers generate sounds that acoustically correlate to each other. Even when feedback sound components are input to the microphone via different feedback transmission paths, the feedback sound components may be too highly correlated to be distinguished from each other. In such case, it is difficult to determine which feedback transmission path transmits feedback sound components corresponding to the residual signal input to the adaptive filter. Consequently, it is difficult to stably configure the filter coefficient simulating each feedback transmission path.  
         [0015]     According to the above-mentioned embodiment of the present invention, the correlation reduction process portion decreases the correlation among output sound signals output to the multiple speakers. Each of the speakers and adaptive filters is supplied with the output sound signal processed by the correlation reduction process portion. This makes it possible to decrease the correlation among feedback sound components input to the microphone via different feedback transmission paths. Consequently, it is possible to prevent the feedback sound components from being too highly correlated to be distinguished from each other.  
         [0016]     (3) According to the present invention, the above-mentioned howling canceler apparatus is provided with another correlation reduction process portion which generates a difference signal by subtracting the output sound signals from each other and a sum signal by adding the output sound signals with each other, wherein the adaptive filter performs a cross spectrum operation using the sum signal and the difference signal to calculate an estimated error between the transfer function of the corresponding feedback transmission path and the simulated transfer function estimated by the adaptive filter itself, and sets the filter coefficient using this estimated error.  
         [0017]     According to the above-mentioned configuration of the present invention, the correlation reduction process portion generates a difference signal and a sum signal of output sound signals to be output to the speakers. The speakers are supplied with output sound signals before being processed in the correlation reduction process portion. If the speaker is supplied with the output sound signal processed in the correlation reduction process portion, the speaker may generate a sound whose quality is acoustically degraded. According to the present invention, the speaker is supplied with a signal before being processed in the correlation reduction process portion, making it possible to effectively prevent the acoustic sound quality from being degraded.  
         [0018]     On the other hand, the adaptive filter is supplied with a sum signal and a difference signal generated in the correlation reduction process portion. The adaptive filter performs a cross spectrum operation using the sum signal and the difference signal. This operation calculates an estimated error between the transfer function of the corresponding feedback transmission path and the simulated transfer function estimated by the adaptive filter itself. The estimated error is used to calculate the filter coefficient. Accordingly, it is possible to stably set the filter coefficient even when high correlation between sounds generated from the speakers may increase the correlation among feedback sound components input to the microphone via different feedback transmission paths.  
         [0019]     (4) In the above-mentioned howling canceler apparatus, according to the present invention, the adaptive filter is supplied with the output sound signal before being processed in the correlation reduction process portion, and convolutes this supplied output sound signal with the filter coefficient to generate the simulation signal.  
         [0020]     According to the above-mentioned configuration of the present invention, the adaptive filter convolutes the filter coefficient with the output sound signal before being processed in the correlation reduction process portion. In this manner, the filter coefficient is used to convolute with the sound signal input to each speaker. It is possible to more precisely approximate the simulation signal to the feedback sound than the configuration where the filter coefficient is used to convolute with a sum signal and a difference signal.  
         [0021]     (5) Preferably, the inventive howling canceler apparatus further comprises a plurality of delays provided in correspondence to the plurality of the adaptive filters, each delay delaying the output sound signal by a delay time and feeding the delayed output sound signal to the corresponding adaptive filter, the delay time representing a delay time of the feedback sound traveling through the corresponding feedback transmission path.  
         [0022]     According to the present invention, the adaptive filter simulates the transfer function for one feedback transmission path even when the microphone is supplied with input sound signals via multiple feedback transmission paths. This makes it possible to provide the sound amplification system simulating each transfer function for each feedback transmission path in comparison with the conventional technology that simulates multiple feedback transmission paths using a common adaptive filter. When the adaptive filter outputs a simulation signal, it is subtracted from the input sound signal. Accordingly, feedback sound components can be fully removed from the input sound signal. It is possible to effectively prevent the howling from occurring. 
     
    
     BRIEF DESCRIPTION OF THE DRAWINGS  
       [0023]      FIG. 1  is a block diagram showing the outline configuration of a sound amplification system according to the first embodiment.  
         [0024]      FIG. 2  is a block diagram showing the outline configuration of a sound amplification system according to the second embodiment.  
         [0025]      FIG. 3  is a block diagram showing the outline configuration of a sound amplification system according to the third embodiment.  
         [0026]      FIG. 4  is a block diagram showing the outline configuration of a sound amplification system according to the fourth embodiment. 
     
    
     DETAILED DESCRIPTION OF THE INVENTION  
       [0027]     Embodiments of the present invention will be described in further detail with reference to the accompanying drawings. In the sound amplification system according to the embodiments, multiple speakers and multiple microphones are connected. Accordingly, the microphones are supplied with a feedback sound output from each of the multiple speakers, i.e., the mixture of multiple feedback sounds fed back through multiple feedback transmission paths. According to the embodiments, a howling canceler apparatus is provided with a delay portion and an adaptive filter corresponding to each of the multiple feedback transmission paths to stably simulate the delay time and the transfer function for each feedback transmission path.  
       FIRST EMBODIMENT  
       [0028]     With reference to  FIG. 1 , the following describes a first embodiment of the present invention.  FIG. 1  is a block diagram showing the outline configuration of a sound amplification system  1  according to the first embodiment. The sound amplification system  1  connects with two (multiple) microphones  2  and two (multiple) speakers  3 . Each microphone  2  is provided with a head amplifier  4  and a mixer  5 . Each speaker  3  is provided with a power amplifier  6  and a howling canceler apparatus  7 . The head amplifier  4 , mixer  5  and power amplifier  6  may collectively or individually constitute a sound amplification portion of the inventive sound amplification system.  
         [0029]     The microphone  2  receives the sound as a microphone input signal from the outside of the apparatus and supplies this microphone input signal to the sound amplification system  1 . Of the two microphones  2  in  FIG. 1 , the left thereof is a microphone  21  the right thereof is a microphone  22 . The following description simply denotes the microphone  2  when there is no need for special distinction between the microphones  21  and  22 .  
         [0030]     The speaker  3  converts the analog sound signal input from the sound amplification system  1  and generates the sound. Of the two speakers in  FIG. 1 , the left thereof is a speaker  31  that works as a first channel to generate the sound. The right thereof is a speaker  32  that works as a second channel to generate the sound. The following description simply denotes the speaker  3  when there is no need for special distinction between the speakers  31  and  32 .  
         [0031]     The speakers  31  and  32  and the microphones  21  and  22  are positioned so that the sound generated from the speakers  31  and  32  is input as a feedback sound to each of the microphones  21  and  22  via a feedback transmission path  100  ( 101 ,  102 ,  103 , and  104 ). That is, the sound generated from the speaker  31  is input to not only the microphone  21  via the feedback transmission path  101 , but also the microphone  22  via the feedback transmission path  102 . The sound generated from the speaker  32  is input to not only the microphone  21  via the feedback transmission path  103 , but also the microphone  22  via the feedback transmission path  104 . In this manner, the microphone  2  is supplied with the feedback sound via multiple types of the feedback transmission path  100 .  
         [0032]     The head amplifier  4  ( 41  and  42 ) is supplied with the microphone input signal from the microphone  2  via an input terminal  8 . The head amplifier  4  amplifies the signal level of the supplied microphone input signal so as to be appropriate to processes for an A/D (Analog/Digital) converter (not shown). The head amplifier  4  inputs the microphone input signal to the A/D converter (not shown). Of the head amplifier  4 , a head amplifier  41  is supplied with the microphone input signal from the microphone  21 . A head amplifier  42  is supplied with the microphone input signal from the microphone  22 . The microphone input signal is amplified in the head amplifiers  41  and  42 , digitized in the A/D converter (not shown), and output to a mixer  5 .  
         [0033]     The mixer  5  mixes and possibly preamplifies input signals. The mixer  5  is supplied with the microphone input signals output from the head amplifiers  41  and  42  via the howling canceler apparatus  7 . The mixer mixes these input signals to generate sound signals x 1 (k) and x 2 (k). The mixer outputs the sound signal x 1 (k) to the speaker  31  and outputs the sound signal x 2 (k) to the speaker  32 . The output sound signals x 1 (k) and x 2 (k) are input to not only the power amplifier  6 , but also the howling canceler apparatus  7 . In this manner, the howling canceler apparatus  7  is supplied with the same signals as the sound signals x 1 (k) and x 2 (k) input to the speaker  3 . According to this configuration, the howling canceler apparatus  7  is supplied with the sound signals x 1 (k) and x 2 (k) that do not pass through the power amplifier  6 . According to another configuration, the howling canceler apparatus  7  may be supplied with the sound signals x 1 (k) and x 2 (k) that pass through the power amplifier  6 .  
         [0034]     The power amplifier  6  corresponds to the sound amplification portion in the present invention. The power amplifier  6  amplifies signal levels of the input sound signals x 1 (k) and x 2 (k) and outputs them to the speaker  3 . Two power amplifiers  6  are provided. Of these, a power amplifier  61  outputs signals to the speaker  31 . A power amplifier  62  outputs signals to the speaker  32 . Signals output from the power amplifiers  61  and  62  are respectively input to the speakers  31  and  32  via an output terminal  9 . The power amplifiers  61  and  62  may be digital amplifiers for amplifying digital signals or analog amplifiers for amplifying analog signals. When the analog amplifiers are used, a D/A converter (not shown) is placed previously to the power amplifiers  61  and  62 .  
         [0035]     The howling canceler apparatus includes a delay portion  71  ( 711 ,  712 ,  713 , and  714 ) an adaptive filter  72  ( 721 ,  722 ,  723 , and  724 ), an addition portion  73  ( 731  and  732 ), and a subtraction portion  74  ( 741  and  742 ).  
         [0036]     The delay portion  71  and the adaptive filter  72  simulates the feedback transmission path  100  that forms a sound transmission route from the speaker  3  to the microphone  2 . That is, the delay portion  71  simulates delay time τ of the feedback sound via the feedback transmission path  100 . The adaptive filter  72  simulates transfer function h, i.e., the audio propagation characteristic of the feedback transmission path  100 . Multiple delay portions  71  and adaptive filters  72  are provided for each of the feedback transmission path  100 . That is, the delay portion  711  and the adaptive filter  721  simulate the feedback transmission path  101 . The delay portion  712  and the adaptive filter  722  simulate the feedback transmission path  103 . The delay portion  713  and the adaptive filter  723  simulate the feedback transmission path  102 . The delay portion  714  and the adaptive filter  724  simulate the feedback transmission path  104 .  
         [0037]     Specifically, the delay portion  71  delays the input sound signals x 1 (k) and x 2 (k) for delay time τ that simulates the delay time of the feedback transmission path  100 . The delay portion  71  outputs this delayed sound signal x(k-τ) to the adaptive filter  72  that simulates the same feedback transmission path  100  as itself. That is, the delay portion  711  delays sound signal x 1 (k) for delay time τ 11  to simulate the delay time of the feedback transmission path  101  and outputs delayed sound signal x 1 (k-τ 11 ) to the adaptive filter  721 . The delay portion  712  delays sound signal x 2 (k) for delay time τ 21  of the feedback transmission path  103  and outputs delayed sound signal x 2 (k-τ 21 ) to the adaptive filter  722 . The delay portion  713  delays sound signal x 1 (k) for delay time τ 12  of the feedback transmission path  102  and outputs delayed sound signal x 1 (k-τ 12 ) to the adaptive filter  723 . The delay portion  714  delays sound signal x 2 (k) for delay time τ 22  of the feedback transmission path  104  and outputs delayed sound signal x 2 (k-τ 22 ) to the adaptive filter  724 . This specification simply describes delay time “τ” when there is no need for special distinction between delay times τ 11 , τ 21 , τ 12 , and τ 22 .  
         [0038]     The adaptive filter  72  includes a digital filter (typically an FIR (Finite Impulse Response) filter). The adaptive filter  72  estimates transfer function h of the feedback transmission path  100 . The adaptive filter  72  calculates this digital filter&#39;s filter coefficient (filter characteristic) so as to adjust to (or simulate) the estimated transfer function h and assigns the filter coefficient to itself. The adaptive algorithm is used to estimate transfer function h and calculate the filter coefficient using, as a reference signal, the residual signal output from the subtraction portion  74  based on sound signal x(k-τ) input from the delay portion  71 . Applicable adaptive algorithms include the learning identification method, the LMS method, the projection method, and the RLS method, for example. The filter coefficient is calculated at a specified time interval (e.g., every several seconds) so as to generate as small a residual signal as possible. The adaptive filter  72  generates simulation signal do(k) by convoluting the input sound signal x 1 (k-τ) or x 2 (k-τ) with the filter coefficient (thus, providing the filter characteristic). The adaptive filter  72  outputs generated simulation signal do(k) to the addition portion  73 .  
         [0039]     The adaptive filter  721  simulates transfer function h 11  for the feedback transmission path  101 , generates simulation signal do 1 (k) by convoluting the input sound signal x 1 (k-τ 11 ) with the filter coefficient, and outputs generated simulation signal do 1 (k) to the addition portion  73  (addition portion  731 ). The adaptive filter  722  simulates transfer function h 21  for the feedback transmission path  103 , generates simulation signal do 2 (k) by convoluting the input sound signal x 2 (k-τ 21 ) with the filter coefficient, and outputs generated simulation signal do 2 (k) to the addition portion  73  (addition portion  731 ). The adaptive filter  723  simulates transfer function h 12  for the feedback transmission path  102 , generates simulation signal do 3 (k) by convoluting the input sound signal x 1 (k-τ 12 ) with the filter coefficient, and outputs generated simulation signal do 3 (k) to the addition portion  73  (addition portion  732 ). The adaptive filter  724  simulates transfer function h 22  for the feedback transmission path  104 , generates simulation signal do 4 (k) by convoluting the input sound signal x 2 (k-τ 22 ) with the filter coefficient, and outputs generated simulation signal do 4 (k) to the addition portion  73  (addition portion  732 ). This specification simply describes simulation signal do(k) when there is no need for special distinction between simulation signals do 1 (k), do 2 (k), do 3 (k), and do 4 (k).  
         [0040]     The addition portion  73  synthesizes simulation signals do(k) with each other. Two (multiple) addition portions  73  are respectively provided for the microphones  21  and  22 . The addition portion  731  of the addition portion  73  corresponds to the microphone  21 . The addition portion  732  of the addition portion  73  corresponds to the microphone  22 . The addition portion  731  is supplied with simulation signals do 1 (k) and do 2 (k). The addition portion  731  adds these signals to generate synthesized simulation signal do 10 (k), thus generating a signal simulating the feedback sound supplied to the microphone  21 . The addition portion  732  is supplied with simulation signals do 3 (k) and do 4 (k). The addition portion  732  adds these signals to generate synthesized simulation signal do 20 (k), thus generating a signal simulating the feedback sound supplied to the microphone  22 .  
         [0041]     The microphone  21  is supplied with synthesized simulation signal d 10 (k) of feedback sound signals d 1 (k) and d 2 (k). The feedback sound d 1 (k) corresponds to the feedback sound via the feedback transmission path  101 . The feedback sound d 2 (k) corresponds to the feedback sound via the feedback transmission path  103 . The microphone  22  is supplied with synthesized simulation signal d 20 (k) of feedback sound signals d 3 (k) and d 4 (k). The feedback sound d 3 (k) corresponds to the feedback sound via the feedback transmission path  102 . The feedback sound d 4 (k) corresponds to the feedback sound via the feedback transmission path  104 . Since the adaptive filter  721  simulates transfer function h 11  as mentioned above, simulation signal do 1 (k) simulates feedback sound signal d 1 (k). Since the adaptive filter  722  simulates transfer function h 21  as mentioned above, simulation signal do 2 (k) simulates feedback sound signal d 1 (k). Accordingly, synthesized simulation signal d 10 (k) approximates to simulation signal do 10 (k). Since the adaptive filter  723  simulates transfer function h 12  as mentioned above, simulation signal do 3 (k) simulates feedback sound signal d 3 (k). Since the adaptive filter  724  simulates transfer function h 22  as mentioned above, simulation signal do 4 (k) simulates feedback sound signal d 4 (k). Accordingly, synthesized simulation signal d 20 (k) approximates to simulation signal do 20 (k). This specification simply describes feedback sound signal d(k) when there is no need for special distinction between feedback sound signals d 1 (k), d 2 (k), d 3 (k), and d 4 (k).  
         [0042]     The addition portion  731  inputs generated synthesized simulation signal do 10 (k) to the subtraction portion  74  (subtraction portion  741  to be described later) corresponding to the microphone  21 . The addition portion  732  inputs generated synthesized simulation signal do 20 (k) to the subtraction portion  74 (subtraction portion  742  to be described later) corresponding to the microphone  22 . The subtraction portion  74  is supplied with a microphone input signal from the microphone  2 . The subtraction portion  74  subtracts synthesized simulation signal do 10 (k) or do 20 (k) from the input signal. two subtraction portions  74  are respectively provided for the microphones  21  and  22 . The subtraction portion  741  is the subtraction portion  74  corresponding to the microphone  21 . The subtraction portion  742  is the subtraction portion  74  corresponding to the microphone  22 .  
         [0043]     That is, the subtraction portion  741  generates a residual signal by subtracting synthesized simulation signal do 10  from the sound signal input from the microphone  21 . The subtraction portion  742  generates a residual signal by subtracting synthesized simulation signal do 20  from the sound signal input from the microphone  22 . The subtraction portion  741  inputs the generated residual signal to the mixer  5  and to the adaptive filters  721  and  722  as the reference signal. The subtraction portion  742  inputs the generated residual signal to the mixer  5  and to the adaptive filters  723  and  724  as the reference signal.  
         [0044]     The following describes operations of the sound amplification system  1 . When a user speaks, for example, the sound signal such as the user&#39;s voice is input to the microphones  21  and  22 . The microphone input signal supplied to the microphone  21  is input to the head amplifier  41  via the input terminal  8 . The microphone input signal supplied to the microphone  22  is input to the head amplifier  42  via the input terminal  8 . The head amplifiers  41  and  42  amplify signal levels of the supplied microphone input signals. The microphone input signals are then input to the mixer  5  via the subtraction portions  741  and  742 . The mixer  5  mixes the microphone input signals supplied from the microphones  21  and  22  to generate sound signals x 1 (k) and x 2 (k).  
         [0045]     The mixer inputs the generated sound signals x 1 (k) and x 2 (k) not only to the power amplifiers  61  and  62 , but also to the delay portions  711 ,  712 ,  713 , and  714 . That is, sound signal x 1 (k) input to the power amplifier  61  is also input to the delay portions  711  and  713 . Sound signal x 2 (k) input to the power amplifier  62  is also input to the delay portions  712  and  714 . The power amplifiers  61  and  62  amplify signal levels of the input sound signals x 1 (k) and x 2 (k) that are then input to the speakers  31  and  32  via the output terminal  9 .  
         [0046]     The analog signal input to the speaker  31  is transformed into sound that is then generated audibly. The sound is input as feedback sound signal d 1 (k) to the microphone  21  via the feedback transmission path  101 . The sound is also input as feedback sound signal d 3 (k) to the microphone  22  via the feedback transmission path  102 . The analog signal input to the speaker  32  is transformed into sound that is then generated audibly. The sound is input as feedback sound signal d 2 (k) to the microphone  21  via the feedback transmission path  103 . The sound is also input as feedback sound signal d 4 (k) to the microphone  22  via the feedback transmission path  104 . That is, the microphone  21  is supplied with synthesized simulation signal d 10 (k) composed of feedback sound signals d 1 (k) and d 2 (k). The microphone  22  is supplied with synthesized simulation signal d 20 (k) composed of feedback sound signals d 3 (k) and d 4 (k).  
         [0047]     The howling canceler apparatus  7  uses the delay portions  711 ,  712 ,  713 , and  714  to provide delay time τ for sound signals x 1 (k) and x 2 (k). That is, the delay portion  711  provides delay time τ 11  to sound signal x 1 (k) to generate sound signal x 1 (k-τ 11 ) that is then input to the adaptive filter  721 . The delay portion  712  provides delay time τ 21  to sound signal x 2 (k) to generate sound signal x 2 (k-τ 2 l) that is then input to the adaptive filter  722 . The delay portion  713  provides delay time τ 12  to sound signal x 1 (k) to generate sound signal x 1 (k-τ 12 ) that is then input to the adaptive filter  723 . The delay portion  714  provides delay time τ 22  to sound signal x 2 (k) to generate sound signal x 2 (k-τ 22 ) that is then input to the adaptive filter  724 .  
         [0048]     The adaptive filter  721  supplies sound signal x 1 (k-τ 11 ) with the filter characteristic corresponding to the feedback transmission path  101  to generate simulation signal do 1 (k). The generated simulation signal do 1 (k) is input to the addition portion  731 . The adaptive filter  722  supplies sound signal x 2 (k-τ 21 ) with the filter characteristic corresponding to the feedback transmission path  103  to generate simulation signal do 2 (k). The generated simulation signal do 2 (k) is input to the addition portion  731 . The adaptive filter  723  supplies sound signal x 1 (k-τ 12 ) with the filter characteristic corresponding to the feedback transmission path  102  to generate simulation signal do 3 (k). The generated simulation signal do 3 (k) is input to the addition portion  732 . The adaptive filter  724  supplies sound signal x 2 (k-τ 22 ) with the filter characteristic corresponding to the feedback transmission path  104  to generate simulation signal do 4 (k). The generated simulation signal do 4 (k) is input to the addition portion  732 .  
         [0049]     The addition portion  731  adds simulation signals do 1 (k) and do 2 (k) to generate synthesized simulation signal do 10 (k). The synthesized simulation signal do 10 (k) is input to the subtraction portion  741 . The addition portion  732  adds simulation signals do 3 (k) and do 4 (k) to generate synthesized simulation signal do 20 (k). The synthesized simulation signal do 20 (k) is input to the subtraction portion  742 . The subtraction portion  742  removes synthesized simulation signal do 10 (k) from the microphone input signal supplied from the microphone  21  to remove components of synthesized simulation signal d 10 (k). The subtraction portion  742  removes synthesized simulation signal do 20 (k) from the microphone input signal supplied from the microphone  22  to remove components of synthesized simulation signal d 20 (k). This method removes feedback sound components supplied from microphone input signals supplied from the microphones  21  and  22  via multiple feedback transmission paths  100 . It is possible to effectively prevent the howling.  
         [0050]     According to the above-mentioned configuration, the embodiment provides multiple types of adaptive filters  72  even when the same microphone  2  is supplied with the feedback sound via multiple types of feedback transmission paths  100 . In this manner, the delay time is supplied for each feedback transmission path  100  and transfer function h is simulated. It is possible to stably estimate transfer function h. As a result, synthesized simulation signals do 10 (k) and do 20 (k) can be accurately approximated to synthesized simulation signals d 10 (k) and d 20 (k). It is possible to accurately prevent the howling.  
         [0051]     Further, the delay portion  71  is provided for each feedback transmission path  100 . Sound signal x(k) is delayed for delay time τ corresponding to each feedback transmission path  100  and is input to the adaptive filter  72 . It is possible to accurately match the input timing between feedback sound signal d(k) and simulation signal do(k) supplied to the subtraction portion  74 . Since simulation signal do(k) is removed from the simulation signal, it is possible to appropriately remove feedback sound components corresponding to simulation signal do(k). Accordingly, this makes it possible to accurately prevent the howling.  
       SECOND EMBODIMENT  
       [0052]     Referring now to  FIG. 2 , the following describes a sound amplification system  1 A according to a second embodiment of the present invention.  FIG. 2  is a block diagram showing the outline configuration of the sound amplification system  1 A according to the second embodiment of the present invention. According to the first embodiment, the speakers  31  and  32  are supplied with sound signals x 1 (k) and x 2 (k) supplied from the mixer  5  via the power amplifier  6 . The delay portion  71  is supplied with sound signals x 1 (k) and x 2 (k) output from the mixer  5 . By contrast, the second embodiment performs a process (correlation reduction process) to decrease the correlation between sound signals x 1 (k) and x 2 (k). After this process, sound signals x 1 ′(k) and x 2 ′(k) are respectively input to the speakers  31  and  32  via the power amplifier  6  and also to delay portions  711 A and  713 A and  712 A and  714 A.  
         [0053]     In addition to the same configuration as the howling canceler apparatus  7 , the howling canceler apparatus  7 A in  FIG. 2  is provided with a correlation reduction process portion  75 . The correlation reduction process portion  75  is positioned along the signal route between the mixer  5  and the power amplifier  6  and between the mixer  5  and a branch to the delay portion  71 A on this signal route. The correlation reduction process portion  75  is equivalent to a first correlation reduction process portion according to the present invention. The correlation reduction process portion  75  applies a correlation reduction process to sound signals x 1 (k) and x 2 (k) supplied from the mixer  5 . The correlation reduction process portion  75  applies the correlation reduction process to sound signal x 1 (k) to generate sound signal x 1 ′(k) and supplies sound signal x 1 ′(k) to the power amplifier  61  and the delay portions  711 A and  713 A. The correlation reduction process portion  75  applies the correlation reduction process to sound signal x 2 (k) to generate sound signal x 2 ′(k) and supplies sound signal x 2 ′(k) to the power amplifier  62  and the delay portions  712 A and  714 A.  
         [0054]     The correlation reduction process portion  75  performs the following correlation reduction processes, for example. One process supplies one of sound signals x 1 (k) and x 2 (k) with noise components such as white noise as an identification signal. Another process (MS system) generates a sum signal and a difference signal between sound signals x 1 (k) and x 2 (k) and uses them as sound signals x 1 ′(k) and x 2 ′(k), respectively. Yet another process (orthogonalization) analyzes main components of sound signals x 1 (k) and x 2 (k) and transforms these signals into two signals that are orthogonal to each other.  
         [0055]     Similarly to the first embodiment, each delay portion  71 A delays input sound signals x 1 ′(k) and x 2 ′(k) for delay time τ that corresponds to the delay time for each feedback transmission path  100 . In this manner, the delay portion  71 A generates sound signals x 1 ′(k-τ) and x 2 ′(k-τ) and supplies these signals to an adaptive filter  72 A. The adaptive filter  72 A convolutes the input sound signals x 1 ′(k-τ) and x 2 ′(k-τ) with the filter coefficient to generate simulation signal do(k). Similarly to the first embodiment, the adaptive filter  72 A supplies simulation signal do(k) to the addition portions  731  and  732 . The signal processes in the addition portion  73  and the subtraction portion  74  are the same as those in the first embodiment and a description is omitted.  
         [0056]     The adaptive filter  72 A uses the supplied sound signals x 1 ′(k-τ) and x 2 ′(k-τ) and the residual signal to calculate the filter coefficient using the adaptive algorithm similarly to the first embodiment. The calculated filter coefficient is used for correction. That is, an adaptive filter  721 A calculates the filter coefficient using supplied sound signal x 1 (k-τ 11 ) and the residual signal supplied from the subtraction portion  741 . An adaptive filter  722 A calculates the filter coefficient using supplied sound signal x 2 ′(k-τ 21 ) and the residual signal supplied from the subtraction portion  741 . An adaptive filter  723 A calculates the filter coefficient using supplied sound signal x 1 ′(k-τ 12 ) and the residual signal supplied from the subtraction portion  742 . An adaptive filter  724 A calculates the filter coefficient using supplied sound signal x 2 ′(k-τ 22 ) and the residual signal supplied from the subtraction portion  742 .  
         [0057]     When there is close correlation between sounds generated from the speakers  31  and  32 , for example, the correlation increases between feedback sound signals d 1 (k) and d 2 (k) input to the microphone  21 . The correlation also increases between feedback sound signals d 3 (k) and d 4 (k) input to the microphone  22 . For this reason, it is difficult to determine whether the residual signal originates from feedback sound signal d 1 (k) or d 2 (k). Further, it is difficult to determine whether the residual signal originates from feedback sound signal d 3 (k) or d 4 (k). The second embodiment prevents this situation as follows. The correlation reduction process portion  75  applies the correlation reduction process to mixed sound signals x 1 (k) and x 2 (k) to decrease the correlation between them. The sound signals are supplied as x 1 ′(k) and x 2 ′(k) to the speakers  31  and  32 .  
         [0058]     According to the above-mentioned configuration, the second embodiment uses the correlation reduction process portion  75  to supply the speakers  31  and  32  with sound signals x 1 ′(k) and x 2 ′(k) whose correlation is decreased. It is possible to effectively prevent the difficulty in determining whether the residual signal originates from feedback sound components transmitted to which feedback transmission path  100 . An appropriate filter coefficient can be calculated.  
       THIRD EMBODIMENT  
       [0059]     Referring now to  FIG. 3 , the following describes a third embodiment of the present invention.  FIG. 3  is a block diagram showing the outline configuration of a sound amplification system  1 B according to the third embodiment of the present invention. According to the second embodiment, the correlation reduction process portion  75  supplies the delay portion  71 A and the speakers  31  and  32  with sound signals x 1 ′(k) and x 2 ′(k) to which the correlation reduction process is applied. This configuration decreases the correlation between sounds generated from the speakers  31  and  32 . In this manner, it is possible to use the adaptive filter  72 A to stably calculate the filter coefficient. By contrast, the third embodiment supplies the speakers  31  and  32  with sound signals x 1 (k) and x 2 (k) to which no correlation reduction process is applied. This does not decrease the correlation between sounds generated from the speakers  31  and  32 . To solve this problem, a correlation reduction process portion  75 ′ supplies a delay portion  71 B with sound signals x 1 ′(k) and x 2 ′(k) to which the correlation reduction process is applied. Each adaptive filter  72 B performs an estimated error calculation process (to be described) using sound signals x 1 ′(k) and x 2 ′(k) and the residual signal to calculate estimated error Δh between transfer function h for the feedback transmission path  100  and the transfer function estimated by the adaptive filter  72 B itself. The adaptive filter  72 B uses this estimated error Δh to calculate the filter coefficient. Since each adaptive filter  72 B calculates the filter coefficient using estimated error Δh, the filter coefficient can be stably calculated. In this manner, the third embodiment is characterized by stably calculating the filter coefficient while maintaining the quality of generated sound.  
         [0060]     In the sound amplification system  1 B of  FIG. 3 , the correlation reduction process portion  75 ′ is positioned along the signal route between the delay portion  71 B and the branch from the signal route between the mixer  5  and the power amplifier  6 . The correlation reduction process portion  75 ′ uses the MS system as mentioned in the second embodiment to apply the correlation reduction process to sound signals x 1 (k) and x 2 (k) supplied from the mixer  5 . The processed sound signals are input to the delay portion  71 B.  
         [0061]     Specifically, the correlation reduction process portion  75 ′ is composed of a subtractor, an adder, and the like. The MS-based correlation reduction process generates a sum signal (sound signal x 1 ′(k)) of sound signals x 1 (k) and x 2 (k) and a difference signal (sound signal x 2 ′(k)) between sound signals x 1 (k) and x 2 (k), i.e., “x 1 (k)-x 2 (k)” or “x 2 (k)-x 1 (k)”. The correlation reduction process portion  75 ′ supplies sound signals x 1 ′(k) and x 2 ′(k) to the delay portions  711 B,  712 B,  713 B, and  714 B.  
         [0062]     The delay portion  711 B delays sound signals x 1 ′(k) and x 2 ′(k) supplied using delay time τ 11  corresponding to the delay time for each feedback transmission path  100  similarly to the first embodiment to generate sound signals x 1 ′(k-τ 11 ) and x 2 ′(k-τ 11 ) that are then input to the adaptive filter  721 B. The delay portion  712 B delays sound signals x 1 ′(k) and x 2 ′(k) supplied using delay time τ 21  corresponding to the delay time for each feedback transmission path  100  similarly to the first embodiment to generate sound signals x 1 ′(k-τ 21 ) and x 2 ′(k-τ 21 ) that are then input to the adaptive filter  722 B. The delay portion  713 B delays sound signals x 1 ′(k) and x 2 ′(k) supplied using delay time τ 12  corresponding to the delay time for each feedback transmission path  100  similarly to the first embodiment to generate sound signals x 1 ′(k-τ 12 ) and x 2 ′(k-τ 12 ) that are then input to the adaptive filter  723 B. The delay portion  714 B delays sound signals x 1 ′(k) and x 2 ′(k) supplied using delay time τ 22  corresponding to the delay time for each feedback transmission path  100  similarly to the first embodiment to generate sound signals x 1 ′(k-τ 22 ) and x 2 ′(k-τ 22 ) that are then input to the adaptive filter  724 B.  
         [0063]     Each adaptive filter  72 B convolutes the supplied sound signal x 1 ′(k-τ) or k 2 ′(k-τ) with the filter coefficient to generate simulation signal do(k). Specifically, the adaptive filter  721 B convolutes the supplied x 1 ′(k-τ 11 ) with the filter coefficient to generate simulation signal do 1 (k) and supplies it to the addition portion  731  similarly to the first embodiment. The adaptive filter  722 B convolutes the supplied x 2 ′(k-τ 21 ) with the filter coefficient to generate simulation signal do 2 (k) and supplies it to the addition portion  731  similarly to the first embodiment. The adaptive filter  723 B convolutes the supplied x 1 ′(k-τ 12 ) with the filter coefficient to generate simulation signal do 3 (k) and supplies it to the addition portion  732  similarly to the first embodiment. The adaptive filter  724 B convolutes the supplied x 2 ′(k-τ 22 ) with the filter coefficient to generate simulation signal do 4 (k) and supplies it to the addition portion  732  similarly to the first embodiment.  
         [0064]     Each adaptive filter  72 B performs a cross spectrum operation using the supplied sound signals x 1 ′(k-τ) and x 2 ′(k-τ) and the residual signal to calculate estimated error Δh between the transfer function simulated by each adaptive filter  72 B and transfer function h for the corresponding feedback transmission path  100 . Each adaptive filter  72 B uses the calculated estimated error Δh to calculate the filter coefficient and assigns the calculated filter coefficient to itself.  
         [0065]     Specifically, the adaptive filter  721 B uses sound signals x 1 ′(k-τ 11 ) and x 2 ′(k-τ 11 ) and the residual signal supplied from the subtraction portion  741 . The adaptive filter  721 B further uses the following equation to calculate estimated error Δh 11  and uses this estimated error Δh 11  to calculate the filter coefficient. 
 
Estimated error Δ h 11=Σ X 1′*× E   L   /Σ|X 1′| 2   +ΣX 2′*× E   L   /Σ|X 2′| 2   [Equation 1]
 
         [0066]     In this equation, X 1 ′ represents sound signals x 1 ′(k-τ 11 ), x 1 ′(k-τ 21 ), x 1 ′(k-τ 12 ), and x 1 ′(k-τ 22 ) in terms of the frequency axis. X 2 ′ represents x 2 ′(k-τ 11 ), x 2 ′(k-τ 21 ), x 2 ′(k-τ 12 ), and x 2 ′(k-τ 22 ) in terms of the frequency axis. X 1 ′* is the complex conjugate of X 1 ′ and X 2 ′* is the complex conjugate of X 2 ′. E L  represents the residual signal supplied from the subtraction portion  741  in terms of the frequency axis.  
         [0067]     The adaptive filter  722 B uses sound signals x 1 ′(k-τ 21 ) and x 2 ′(k-τ 21 ) and the residual signal supplied from the subtraction portion  741 . The adaptive filter  722 B further uses the following equation to calculate estimated error Δh 21  and uses this estimated error Δh 21  to calculate the filter coefficient. 
 
Estimated error Δ h 21 Σ X 1′*× E   L   /Σ|X 1′| 2   −ΣX 2′*× E   L   /Σ|X 2′| 2   [Equation 2]
 
         [0068]     Specifically, the adaptive filter  723 B uses sound signals x 1 ′(k-τ 12 ) and x 2 ′(k-τ 12 ) and the residual signal supplied from the subtraction portion  742 . The adaptive filter  723 B further uses the following equation to calculate estimated error Δh 12  and uses this estimated error Δh 12  to calculate the filter coefficient. 
 
Estimated error Δ h 12 =ΣX 1 ′*×E   R   /Σ|X 1′| 2   +ΣX 2 ′*×E   R   /Σ|X 2′| 2   [Equation 3]
 
         [0069]     In this equation, E R  represents the residual signal supplied from the subtraction portion  742  in terms of the frequency axis.  
         [0070]     The adaptive filter  724 B uses sound signals x 1 ′(k-τ 22 ) and x 2 ′(k-τ 22 ) and the residual signal supplied from the subtraction portion  742 . The adaptive filter  724 B further uses the following equation (4) to calculate estimated error Δh 22  and uses this estimated error Δh 22  to calculate the filter coefficient. 
 
Estimated error Δ h 22=Σ X 1′*× E   R   /Σ|X 1′| 2   −ΣX 2′*× E   R   /Σ|X 2′| 2   [Equation 4]
 
         [0071]     As disclosed in Japanese Non-examined Patent Publication No. 2003-102085, for example, the known method is used to calculate the filter coefficient using estimated errors Δh 11 ,  12 ,  21 , and  22 , and a description is omitted.  
         [0072]     According to the above-mentioned configuration, the third embodiment performs the cross spectrum operation using the residual signal and sound signals x 1 ′(k-τ) and x 2 ′(k-τ) to which the correlation reduction process portion  75  applies the correlation reduction process. Consequently, it is possible to calculate estimated error Δh between each adaptive filter  72 B and the transfer function for the corresponding feedback transmission path. Estimated error Δh can be used to calculate the filter coefficient for each adaptive filter  72 B. Even when the speakers  31  and  32  generate highly correlated sounds, the filter coefficient can be stably calculated. When the speakers  31  and  32  are supplied with sound signals x 1 (k) and x 2 (k) to which no correlation reduction process is applied, the filter coefficient for the adaptive filter  72 B can be stably calculated. Compared to the second embodiment that supplies the speakers  31  and  32  with sound signals x 1 ′(k) and x 2 ′(k) to which the correlation reduction process is applied, it is possible to prevent deterioration of the quality of sounds generated from the speakers  31  and  32 . In addition, the filter coefficient can be stably calculated.  
         [0073]     The present invention is not limited thereto and may apply the correlation reduction process according to the orthogonal transform as mentioned above in the second embodiment. According to the modification, the correlation reduction process portion  75 ′ is composed of an orthogonalization filter and the like. The correlation reduction process portion  75 ′ analyzes main components of sound signals x 1 (k) and x 2 (k) at a specified time interval and transforms sound signals x 1 (k) and x 2 (k) into two signals that are orthogonal to each other (having phases shifted 90 degrees). The correlation reduction process portion  75 ′ supplies sound signals x 1 ′(k) and x 2 ′(k) to delay portions  711 B,  712 B,  713 B, and  714 B. Similarly to the third embodiment, the delay portion  71 B provides delay time τ for the supplied sound signals x 1 ′(k) and x 2 ′(k) and supplies these signals to the adaptive filter  72 B. The adaptive filters  721 B and  723 B convolute sound signal x 1 ′(k-τ) with the filter coefficient to generate simulation signals do 1 (k) and do 3 (k). The adaptive filters  722 B and  724 B convolute sound signal x 2 ′(k-τ) with the filter coefficient to generate simulation signals do 2 (k) and do 4 (k). Each adaptive filter  72 B calculates estimated error Δh for the transfer function using sound signals x 1 ′(k-τ) and x 2 ′(k-τ) and the residual signal. The specific calculation method complies with the publicly know technology as disclosed in Japanese Non-examined Patent Publication No. 2003-102085, for example, and a description is omitted. The other configurations and signal processes in this modification are the same as those described in the third embodiment and a description is omitted.  
       FOURTH EMBODIMENT  
       [0074]     Referring now to  FIG. 4 , the following describes a sound amplification system  1 C according to a fourth embodiment of the present invention.  FIG. 4  is a block diagram showing the outline configuration of the sound amplification system  1 C according to the fourth embodiment of the present invention. According to the third embodiment, each adaptive filter  72 B uses the filter coefficient to perform the convolution operation for sound signal x 1 ′(k-τ) or x 2 ′(k-τ), i.e., sound signals to which the correlation reduction process is applied. According to the fourth embodiment, each adaptive filter  72 C uses the filter coefficient to perform the convolution operation for sound signal x 1 (k-τ) or x 2 (k-τ).  
         [0075]     The delay portion  75 ′ supplies the delay portion  71 C with not only sound signals x 1 ′(k) and x 2 ′(k), but also sound signal x 1 (k) or x 2 (k). That is, sound signal x 1 (k) is supplied to the delay portions  711 C and  713 C. Sound signal x 2 (k) is supplied to the delay portions  712 C and  714 C. The delay portion  711 C delays supplied sound signals x 1 ′(k), x 2 ′(k), and x 1 (k) for delay time τ 11  and supplies these signals to the adaptive filter  721 C. The delay portion  712 C delays supplied sound signals x 1 ′(k), x 2 ′(k), and x 2 (k) for delay time τ 21  and supplies these signals to the adaptive filter  722 C. The delay portion  713 C delays supplied sound signals x 1 ′(k), x 2 ′(k), and x 1 (k) for delay time τ 12  and supplies these signals to the adaptive filter  723 C. The delay portion  714 C delays supplied sound signals x 1 ′(k), x 2 ′(k), and x 2 (k) for delay time τ 22  and supplies these signals to the adaptive filter  724 C.  
         [0076]     Similarly to the third embodiment, the adaptive filter  72 C calculates the filter coefficient using the supplied sound signals x 1 ′(k-τ) and x 2 ′(k-τ) and the residual signal. The adaptive filter  72 C assigns the calculated filter coefficient to itself. The adaptive filter  72 C generates simulation signal do(k) by convoluting the supplied sound signal x 1 (k-τ) or x 2 (k-τ) with the filter coefficient. Specifically, the adaptive filter  721 C convolutes sound signal x 1 (k-τ 11 ) with the filter coefficient to generate simulation signal do 1 (k) and supplies it to the addition portion  731 . The adaptive filter  722 C convolutes sound signal x 2 (k-τ 21 ) with the filter coefficient to generate simulation signal do 2 (k) and supplies it to the addition portion  731 . The adaptive filter  723 C convolutes sound signal x 1 (k-τ 12 ) with the filter coefficient to generate simulation signal do 3 (k) and supplies it to the addition portion  732 . The adaptive filter  724 C convolutes sound signal x 2 (k-τ 22 ) with the filter coefficient to generate simulation signal do 4 (k) and supplies it to the addition portion  732 . The other configurations and signal processes of the sound amplification system  1 C are the same as those described in the third embodiment and a description is omitted.  
         [0077]     According to the above-mentioned configuration, the fourth embodiment delays sound signals x 1 (k) and x 2 (k) identical to those supplied to the speakers  31  and  32  to generate sound signals x 1 (k-τ) and x 2 (k-τ). The fourth embodiment can convolute these delayed signals with the filter coefficient to generate simulation signal do(k). It is possible to more accurately generate simulation signal do(k) approximate to feedback sound signal d(k). This makes it possible to further improve the accuracy of preventing the howling.  
         [0078]     The embodiments of the present invention can employ the following modifications.  
         [0079]     (1) According to the first through fourth embodiments, the sound amplification systems  1 ,  1 A,  1 B, and  1 C are configured to be attached with the microphone  2  and the speaker  3  externally. The present invention is not limited thereto. The sound amplification systems  1 ,  1 A,  1 B, and  1 C may be integrated with the microphone  2  and the speaker  3 . The sound amplification systems  1 ,  1 A,  1 B, and  1 C include the howling canceler apparatuses  7 ,  7 A,  7 B, and  7 B but may connect with these howling canceler apparatuses externally.  
         [0080]     (2) According to the first through fourth embodiments, the sound amplification systems  1 ,  1 A,  1 B, and  1 C connect with the two microphones  2  and the two speakers  3 . The present invention is not limited thereto. The embodiments only need to connect with the multiple speakers  3  and supply at least one microphone  2  with feedback sounds from the multiple feedback transmission paths  100 . The single microphone  2  may be provided. In this case, the adaptive filters  72 ,  72 A,  72 B, and  72 C are provided for the number of feedback transmission paths  100 . When one microphone  2  and the two speakers  3  are connected, the microphone  2  is normally supplied with feedback sounds via the two feedback transmission paths  100 . Accordingly, there are provided two adaptive filters  72 ,  72 A,  72 B, and  72 C corresponding to the two feedback transmission paths  100 .  
         [0081]     (3) There may be a case where the speaker  3  is distant from the microphone  2  too far to transmit the feedback sound. In such case, it is assumed that there is no feedback transmission path  100 . It may be unnecessary to provide the corresponding adaptive filters  72 ,  72 A,  72 B, and  72 C. With respect to the first embodiment, for example, let us assume that the speaker  31  is distant from the microphone  21  too far to transmit the feedback sound. Since it is assumed that there is no feedback transmission path  101 , the delay portion  711  and the adaptive filter  721  are unneeded.  
         [0082]     (4) The second embodiment provides the correlation reduction process portion  75  independently of the mixer  5 . Further or alternatively, the mixer  5  may have the function of the correlation reduction process portion  75 .  
         [0083]     (5) According to the third and fourth embodiments, the correlation reduction process portion  75 ′ is provided along the signal route from an intermediate branch along the signal route between the mixer  5  and the power amplifier  6 . The present invention is not limited to this configuration. The third embodiment only needs to be configured so that the speakers  31  and  32  can be supplied with sound signals x 1 (k) and x 2 (k), and that the delay portion  71 B can be supplied with sound signals x 1 ′(k) and x 2 ′(k) (sound signals applied with the correlation reduction process). The fourth embodiment only needs to be configured so that the speakers  31  and  32  can be supplied with sound signals x 1 (k) and x 2 (k), and that the delay portion  71 C can be supplied with not only sound signals x 1 ′(k) and x 2 ′(k) (sound signals applied with the correlation reduction process), but also sound signals x 1 (k) and x 2 (k). For example, the correlation reduction process portion  75 ′ may be provided at a connection position similar to the correlation reduction process portion  75  according to the second embodiment. The power amplifier  6  may be preceded by a processing portion that retransforms sound signals x 1 ′(k) and x 2 ′(k) to x 1 (k) and x 2 (k). For example, this processing portion halves (sound signal x 1 ′(k) +sound signal x 2 ′(k)) to find sound signal x 1 (k). The processing portion halves (sound signal x 1 ′(k)−sound signal x 2 ′(k)) to find sound signal x 2 (k).