JP4701962B2 - Regression sound removal device - Google Patents

Description

  The present invention relates to a regression sound removing apparatus for preventing acoustic echoes and howling caused by sounds emitted from a speaker being collected around a microphone, and more particularly to a regression sound removing apparatus using an adaptive filter. is there.

  Conventionally, various apparatuses using an adaptive filter for preventing acoustic echo and howling have been disclosed.

  The echo canceling device of Patent Document 1 includes a plurality of microphones, updates the transfer function of the transmission path from each microphone with the error signal after echo removal, and uses this transfer function to filter coefficients of the FIR filter (adaptive filter) Set.

The echo canceller apparatus of Patent Document 2 includes a plurality of microphones, calculates a pseudo echo path characteristic of each transmission path (echo path) from a plurality of total pseudo echo path characteristics estimated in the past and a transfer function at this time, A new total pseudo echo path characteristic is set from the pseudo echo path characteristic and the current transfer function.
Japanese Patent Laid-Open No. 62-120734 Japanese Patent No. 2938076

  However, in Patent Document 1, since the filter coefficient of the adaptive filter is set using the error signal, it is necessary to continue to update the adaptive filter until the error signal converges, and it takes time to set the filter coefficient. Further, in Patent Document 2, since the current total simulated echo path characteristic is calculated by performing matrix calculation using the previous total simulated echo path characteristic, transfer function, and current transfer function, the coefficient of the adaptive filter is set. A complicated calculation process is required.

  In particular, in recent years, in an acoustic system using a speaker array in which a plurality of speakers are arrayed or a microphone array in which a plurality of microphones are arrayed, the directivity of these speaker arrays and microphone arrays is controlled to Often the environment changes rapidly and non-linearly. In such a situation, the method of setting the current filter coefficient based on the previous error signal and the content of the filter setting as in the above-mentioned patent documents, the setting of the filter coefficient may follow the change in the acoustic environment. This is not possible, and it takes a long time for the adaptive filter to operate stably.

  Therefore, an object of the present invention is to provide a regression sound removing device that effectively removes the regression sound by stably operating the adaptive filter in a short time even when the acoustic environment changes rapidly and nonlinearly. is there.

  A regression sound removing apparatus according to the present invention includes at least a speaker system and a microphone system, a control means for instructing an acoustic environment to both an acoustic environment forming means for realizing one of a plurality of acoustic environments and a regression sound removing means, and a speaker. Regression sound removal means for generating a pseudo-regression sound signal based on an audio signal input to the system and subtracting the pseudo-regression sound signal from a sound collection signal output from the microphone system. The regression sound removing means stores a plurality of adaptive filter parameters set in accordance with each of the plurality of acoustic environments, and based on the acoustic environment instructions when the acoustic environment instructions are given from the control means. The corresponding parameter is read from the storage means, a pseudo regression signal is generated using the read parameter, and the parameter is updated based on the result of subtracting the pseudo regression signal at that time from the previous collected sound signal. And an adaptive filter that generates a pseudo-regression sound signal.

  In this configuration, when the acoustic environment is instructed by the control unit, the acoustic environment forming unit controls the directivity of the speaker system or the microphone system to form a predetermined acoustic environment. The adaptive filter of the regression sound removing means reads out a parameter corresponding to the sound environment instruction content from the storage means, and sets the parameter. Then, the adaptive filter performs a filtering process on the audio signal with the set parameters to generate a pseudo regression sound signal. The regression sound removing means obtains an output signal by subtracting the pseudo regression sound signal from the collected sound signal. Thus, at the time when the acoustic environment is changed, the adaptive filter generates a pseudo-regression sound signal based on a parameter in accordance with the newly set acoustic environment stored in advance in the storage unit. After the initial processing after changing the acoustic environment, the operation of the normal adaptive filter, that is, the operation of generating the pseudo regression sound signal while sequentially updating the parameters to the optimum conditions based on the previous error signal is repeated. .

  Thereby, even if the acoustic environment changes suddenly and nonlinearly, initial parameters suitable for the new acoustic environment can be set immediately, and the optimum parameters can be obtained in a short time.

  In addition, when the adaptive filter of the regression sound removing apparatus according to the present invention receives a new acoustic environment instruction, the adaptive filter updates and stores the currently used parameter in the storage unit, and reads the parameter based on the new acoustic environment instruction. It is characterized by.

  In this configuration, the parameter optimized by the adaptive filter is fed back to the storage means and stored. As a result, when the same acoustic environment instruction is given next time, the initial parameter setting content becomes closer to the optimum state for the instructed acoustic environment, and the optimum parameters can be obtained in a shorter time. Can do.

  In the regression sound removing apparatus of the present invention, the speaker system is a speaker array, the acoustic environment is set according to the directivity of the speaker, the directivity of the speaker array is changed according to the acoustic environment instruction, and the parameters of the adaptive filter are switched. It is characterized by that.

  In this configuration, the parameters of the adaptive filter are stored corresponding to the directivity of the speaker array, and the parameters are read out and set in the adaptive filter based on the directed directivity of the speaker array.

  In the regression removal apparatus of the present invention, the microphone system is a microphone array, the acoustic environment is set according to the directivity of the microphone, the directivity of the microphone array is changed according to the acoustic environment instruction, and the parameters of the adaptive filter are switched. It is characterized by that.

  In this configuration, the parameters of the adaptive filter are stored in correspondence with the directivity of the microphone array, and the parameters are read out and set in the adaptive filter based on the directed directivity of the microphone array.

  In the regression sound removing apparatus of the present invention, the speaker system is a speaker array, the microphone system is a microphone array, the acoustic environment is set by the directivity of the speaker and the directivity of the microphone, and the speaker is set according to the acoustic environment instruction. It is characterized by changing the directivity of the array and the directivity of the microphone array and switching the parameters of the adaptive filter.

  In this configuration, the parameters of the adaptive filter are stored in correspondence with the directivity between the speaker array and the microphone array, and the parameters are read out based on the directivity between the designated speaker array and the microphone array. Set to filter.

  According to the present invention, since the parameters suitable for the designated acoustic environment are set in the adaptive filter at the initial stage of the change, the adaptive filter can be obtained in a short time even if control for changing the acoustic environment rapidly and nonlinearly is performed. Can be operated stably.

A regression sound removing apparatus according to a first embodiment of the present invention will be described with reference to FIGS. In the present embodiment, an echo canceller will be described as an example of a regression sound removing device.
FIG. 1 is a block diagram showing the main part of the echo canceller of this embodiment.
FIG. 2 is a conceptual diagram of filter parameters stored in the memory 13 shown in FIG.
FIG. 3 is a flowchart showing an echo cancellation processing flow of the echo canceller of the present embodiment.

  The echo canceller according to the present embodiment includes an echo cancel unit 1, a speaker unit 3, a microphone unit 4, a sound collection directivity control unit 5, a control unit 7, and an operation input unit 8.

  The control unit 7 controls the entire echo canceller and, based on the acoustic environment setting received by the operation input unit 8, transmits acoustic environment instruction data to the sound collection directivity control unit 5 and the adaptive type of the echo cancellation unit 1. The filter 11 is given. The operation input unit 8 includes operators such as a plurality of buttons, receives various setting inputs from the user, and gives them to the control unit 7.

  The speaker unit 3 is composed of a single speaker and converts a received audio signal to emit sound. The microphone unit 4 includes a microphone array in which a plurality of microphones are arrayed. The microphone unit 4 collects external sounds including a caller's call sound with each microphone and outputs the collected sound to the sound collection directivity control unit 5.

  The sound collection directivity control unit 5 delays and adds the output signals from the microphones of the microphone array based on the acoustic environment instruction data given from the control unit 7, and generates a sound collection signal having sound collection directivity in a predetermined direction. Generate.

  The echo cancellation unit 1 includes an adaptive filter 11, an adder (subtractor) 12, and a memory 13. The adaptive filter 11 includes an FIR filter. By setting a delay coefficient and a filter coefficient of the FIR filter to predetermined values, a pseudo echo (regression) is used by using an impulse response with respect to a received voice signal input from the voice signal input terminal 2. Sound) signal. The adder 12 subtracts the pseudo echo signal from the sound collection signal input from the sound collection directivity control unit 5 and outputs the result. This output signal becomes an error signal and a transmission audio signal, the transmission audio signal is transmitted to the other party via the audio signal output terminal 6, and the error signal is fed back to the adaptive filter 11. As shown in FIG. 2A, the memory 13 stores filter parameters in advance for each sound collection directivity. Specifically, the filter parameter is set for each sound collection directivity set by the microphone unit 4 and the sound collection directivity control unit 5, and is configured by the delay coefficient and filter coefficient of the FIR filter of the adaptive filter 11. Is done. For example, as shown in FIG. 2, if there are sound collection directivities A and B to M, the filter parameters are a0 and b0 to m0 corresponding to the sound collection directivities A and B to M, respectively. To do. Detailed delay coefficients and filter coefficients are set for each of the filter parameters a0, b0 to m0.

  Next, the operation of the adaptive filter 11 will be specifically described along the flowchart of FIG.

When the user operates the operation input unit 8 to set the acoustic environment, the control unit 7 generates acoustic environment instruction data and gives it to the adaptive filter 11.
When the acoustic environment instruction data is input from the control unit 7 (S101), the adaptive filter 11 receives the acoustic environment instruction data and identifies the designated sound collection directivity (S102).

  The adaptive filter 11 reads each delay coefficient and filter coefficient currently set in the FIR filter, and writes them in the memory 13 as filter parameters corresponding to the corresponding sound collection directivity (S103). At this time, the filter parameters of the previous (initial state or the state of the previous update) are stored in the memory 13, but the adaptive filter 11 adds new filter parameters to the already stored filter parameters. Overwrite. For example, in the initial state as shown in FIG. 2A, the filter parameter stored for the sound collection directivity B is b0, but the filter parameter b1 for the sound collection directivity B is included in the adaptive filter 11. If it exists, the adaptive filter 11 overwrites the filter parameter b1 with the filter parameter b1.

  The adaptive filter 11 reads out the filter parameter corresponding to the identified sound collection directivity when the filter parameter set in itself is written in the memory 13 (S104). Then, the adaptive filter 11 sets the delay coefficient and filter coefficient of the FIR filter based on the read filter parameter (S105).

  The adaptive filter 11 performs a convolution operation or a multiplication process on the input received audio signal with a delay coefficient and a filter coefficient (impulse response) set based on the acoustic environment instruction data, and generates a pseudo echo signal. (S106). The adder 12 subtracts the pseudo echo signal from the collected sound signal as described above and outputs the result.

  In this way, simultaneously with the change of the acoustic environment, the filter parameters according to the new acoustic environment stored in the memory 13 are read out and used, so that the filter suitable for the acoustic environment can be used from the initial state after the acoustic environment is changed. Since the parameters are obtained, the filter parameters of the adaptive filter 11 can be optimized in a short time. Thereby, stable echo cancellation can be realized in a short time.

  The adaptive filter 11 receives the error signal generated by the subtraction process of the adder 12 (S107), and calculates and sets the optimum filter parameters at that time using a known learning identification method or the like (S108). . If there is no input of acoustic environment instruction data, the adaptive filter 11 generates a pseudo echo signal using this optimized filter parameter (S108 → S101 → S106).

  The generation of the pseudo echo signal, the input of the error signal, and the calculation / setting of the optimum filter parameters (S106 → S107 → S108) are the operations of the normal adaptive filter 11 and are continuously performed unless the acoustic environment instruction data is input. Executed. As a result, the filter parameters are updated as needed, and asymptotically approach the optimum filter parameters.

  When the acoustic environment instruction data is input, the adaptive filter 11 overwrites and stores in the memory 13 filter parameters that are more optimized for the current acoustic environment. By performing such processing, when the same acoustic environment is set next time, the optimized filter parameter of this time can be used. Can be made. Thereby, stable echo cancellation can be realized in a shorter time.

Next, a regression sound removing apparatus according to the second embodiment will be described with reference to FIGS. In the present embodiment, an echo canceller will be described as an example of a regression sound removing device.
FIG. 4 is a block diagram showing the main part of the echo canceller of this embodiment.
FIG. 5 is a conceptual diagram of filter parameters stored in the memory 13 shown in FIG.
The echo canceller shown in FIG. 4 is configured by a speaker array in which the speaker unit 3 is formed by arranging a plurality of speakers as compared with FIG. 1, and the sound emission directivity control unit 9 is provided between the echo cancellation unit 1 and the speaker unit 3. Is inserted. Further, the echo canceller shown in FIG. 4 gives sound environment instruction data from the control unit 7 to the sound collection directivity control unit 5 as well as to the sound emission directivity control unit 9.

  In such an echo canceller, when the acoustic environment setting is input, the control unit 7 gives the acoustic environment instruction data to the sound emission directivity control unit 9 and the sound collection directivity control unit 5. The sound emission directivity control unit 9 performs delay control of the audio signal output to each speaker of the speaker array based on the sound environment instruction data, and controls the directivity of the sound emitted from the speaker unit 3. The sound collection directivity control unit 5 delay-controls the output signal from each microphone of the microphone array, and generates a sound collection signal having sound collection directivity in a predetermined direction. As described above, the speaker unit 3 is configured by a speaker array, the microphone unit 4 is configured by a microphone array, and the sound emission directivity control unit 9 and the sound collection directivity control unit 5 are provided, so that a wider variety of acoustic environments can be obtained. Can be realized.

  As shown in FIG. 5, the memory 13 stores filter parameters for each combination of sound collection directivity and sound emission directivity. For example, if the sound collection directivity exists only for A and B to M and the sound emission directivity exists for α and β to ρ, Aα0 to Aρ0, Bα0 to Bρ0,. , Mα0 to Mρ0 filter parameters are set and stored.

When the acoustic environment instruction data is input from the control unit 7, the adaptive filter 11 analyzes the acoustic environment instruction data and detects a combination of the corresponding sound collection directivity and sound emission directivity. The adaptive filter 11 reads the corresponding filter parameter, and sets the delay coefficient and filter coefficient of the FIR filter.
Since other operation processes of the adaptive filter 11 are the same as those in the first embodiment, description thereof is omitted.

  As described above, even an acoustic environment in which both the sound emission directivity and the sound collection directivity can be set, that is, an acoustic environment in which various settings can be set as compared with the first embodiment, is stored in the memory. Since the filter parameters need only be read and set, the adaptive filter can be optimized in a short time according to the set acoustic environment, and stable echo cancellation can be realized.

  In particular, when there are a wide variety of acoustic environments as in the present embodiment, by using the configuration of the present invention, stable echo cancellation can be effectively realized in a shorter time than in the past.

Next, a regression sound removing apparatus according to a third embodiment will be described with reference to FIG. This embodiment is different in the filter parameter storage and setting method of the apparatus shown in the second embodiment, and the other configurations are the same. Therefore, the description of the same part of the configuration is omitted.
FIG. 6 is a conceptual diagram of filter parameters stored in the memory of the echo canceller of this embodiment.
In the echo canceller of this embodiment, filter parameters are set in advance and stored in the memory 13 only for the combination of the sound collection directivity and sound emission directivity that are determined to be used in advance (see FIG. 6 (A).)

The echo canceller reads the filter parameter corresponding to this combination when the stored combination of the sound collection directivity and the sound emission directivity (acoustic environment setting) is instructed, and sends it to the adaptive filter 11. Set.

  By using such a configuration, the number of combinations of sound collection directivity / sound emission directivity and filter parameters stored in the memory 13 can be reduced as much as possible, and memory resources are saved. be able to. It should be noted that such filter parameter storage / setting method can also be performed to update and store filter parameters as described above.

  By the way, when setting such filter parameters, a user may instruct a combination of sound collection directivity and sound emission directivity that has not been previously set and stored. In this case, the echo canceller may set the filter parameter of the adaptive filter 11 by any of the following methods.

(1) General-purpose filter parameters that are not related to the combination of the sound collection directivity and sound output directivity are stored.
(2) The filter parameters before the acoustic environment is set by the user are continuously used.
(3) A similar combination is detected from the already stored combinations of the sound collection directivity and sound emission directivity with respect to the designated combination of sound collection directivity and sound output directivity. A filter parameter corresponding to a combination of sound directivity and sound directivity is used. For example, each sound collection directivity and each sound emission directivity are converted into IDs based on the respective directivity features, and similar IDs are selected from the directivity features newly set and detected by the user. This is realized.

Furthermore, the echo canceller of the present embodiment may be provided with the following filter parameter learning function.
When the echo canceller is instructed to store a combination of sound collection directivity and sound emission directivity that is not stored, the echo canceller secures an area in the memory 13 for storing a filter parameter for the combination of sound collection directivity and sound output directivity. (See FIG. 6B.)

  Thereafter, the adaptive filter 11 operates as in the above-described embodiment, and the filter coefficient is updated. When a different acoustic environment is set by the user, the adaptive filter 11 stores the latest filter parameter set for itself in a corresponding area (the newly reserved area described above) of the memory 13 (FIG. 6). (See (C).)

  With this configuration, the added combination of sound collection directivity / sound emission directivity and filter parameters are stored, and the added combination of sound collection directivity / sound release directivity is instructed again. In this case, an optimum filter coefficient can be obtained in a short time.

  As a new filter parameter storage method, for example, when an area corresponding to a new filter parameter is secured in the memory 13, for example, the sound collection directivity / sound output directivity with the least number of uses or the shortest use time is used. There is also a method of eliminating the combination of sex and the filter parameter. In this case, the memory 13 stores the combination of the sound collection directivity / sound output directivity and the filter parameters, and the accumulated number of times and the use time. The adaptive filter 11 reads the number of times of use and time of use, orders the combinations of the sound collection directivity / sound emission directivity and the filter parameters, and deletes the lowest set. Then, a new combination of sound collection directivity / sound emission directivity and a set of filter parameters are stored in the region formed by this processing.

  With such a configuration, memory resources are saved and filter parameters that are easy to use are stored. Therefore, an echo canceller that is easy to use can be realized with a limited memory.

  In each of the above-described embodiments, the case where there is one transmission line of the received audio signal is shown. However, as shown in FIG. 7, even when there are a plurality of transmission lines (three lines) on the sound emission side, By applying the above-described configuration, the above-described effects can be achieved.

FIG. 7 is a block diagram showing a main part of an echo canceller having another configuration.
The echo canceller shown in FIG. 7 has three transmission lines for received voice signals, and is configured by a speaker array, for example, by performing delay control and amplitude control of each received voice signal by the sound emission directivity control unit 9. The speaker system 3 realizes a plurality of virtual point sound sources. In the echo canceller shown in FIG. 7, the microphone unit 4 includes only a single microphone, and the sound collection directivity control unit 5 is omitted.

  In the case of such a configuration, the adaptive filter 11 is provided with three function units corresponding to each channel, and each function unit generates a pseudo echo signal for each received audio signal of each channel. At this time, the filter parameters are stored and set in the memory 13 for each sound output directivity corresponding to each received audio signal.

  Note that the microphone unit 4 may be configured by a microphone array and a sound collection directivity control unit may be provided. In this case, a combination of sound emission directivity and sound collection directivity for each received sound signal. A filter parameter is stored and set for each time.

  In the example shown in FIG. 7, a case where a plurality of virtual point sound sources is realized is shown, but the configuration of the present invention can also be applied to a case where a plurality of speakers are actually installed to emit sound. Furthermore, if the acoustic space (room size, shape, etc.) is variable as well as the speaker unit and microphone unit, the above configuration can be applied by setting the filter parameters including these. Can do.

  In the above description, the coefficients of the adaptive filter are switched in accordance with the sound emission directivity of the speaker array and the sound collection directivity of the microphone array. However, each embodiment of the present invention controls directivity by the array. It is not limited to. For example, the present invention can be applied to a single speaker unit and microphone unit as long as the installation direction can be controlled and detected.

  In the above description, the echo canceller has been described. However, if the sound emitted from the speaker is collected around the microphone (returned), the configuration of the present invention is applied. The effects described above can be achieved. An example of this is a howling canceller.

  In the above description, the filter parameter optimized by the adaptive filter 11 is overwritten in the memory 13, but this processing is not performed, and the filter preset in the memory 13 is received every time the acoustic environment instruction data is received. The parameter may be used every time.

It is a block diagram which shows the principal part of the echo canceller of 1st Embodiment. It is a conceptual diagram of the filter parameter memorize | stored in the memory 13 shown in FIG. It is a flowchart which shows the echo cancellation processing flow of the echo canceller of 1st Embodiment. It is a block diagram which shows the principal part of the echo canceller of 2nd Embodiment. It is a conceptual diagram of the filter parameter memorize | stored in the memory 13 shown in FIG. It is a conceptual diagram of the filter parameter memorize | stored in the memory of the echo canceller of 3rd Embodiment. It is a block diagram which shows the principal part of the echo canceller of another structure.

Explanation of symbols

1-echo canceller, 11-adaptive filter, 12-adder, 13-memory, 2-audio signal input terminal, 3-speaker unit, 4-microphone unit, 5-sound collection directivity control unit, 6-audio signal Output terminal, 7-control unit, 8-operation input unit, 9-sound emission directivity control unit

Claims (4)

A speaker array comprising a plurality of speakers;
With a microphone,
Directivity setting means for setting the directivity of the speaker array;
An adaptive filter that generates a pseudo-regression sound signal based on an audio signal input to the speaker array;
Regression sound removal means for subtracting the pseudo-regression sound signal from the sound collection signal output from the microphone;
Storage means for storing the directivity setting and the adaptive filter parameter in association with each other;
Accepting means for accepting the setting of the directivity;
Based on the directivity setting accepted by the accepting means, the directivity setting means and a control means for instructing the adaptive filter to set directivity,
The adaptive filter reads a parameter corresponding to the setting of directivity instructed from the control unit from the storage unit, generates the pseudo regression sound signal using the read parameter, and then continues the previous sound collection signal. Generating a pseudo regression sound signal while updating the parameter based on the result of subtracting the pseudo regression sound signal at that time from
When an instruction for setting directivity not stored in the storage means is received, the pseudo-regressive sound signal is generated using predetermined parameters, and the pseudo-regressive sound signal at that time is continuously generated from the previous sound collection signal. When the pseudo-regressive sound signal is generated while the parameter is updated based on the subtraction result, and then an instruction for setting a new directivity is received, the undirected directivity setting and the time point used at that time are used. A recurring sound removal apparatus that additionally stores the stored parameters in the storage means. A microphone array consisting of multiple microphones;
Speakers,
Directivity setting means for setting the directivity of the microphone array;
An adaptive filter that generates a pseudo-regression sound signal based on an audio signal input to the speaker;
Regression sound removal means for subtracting the pseudo-regression sound signal from the sound collection signal output from the microphone array;
Storage means for storing the directivity setting and the adaptive filter parameter in association with each other;
Accepting means for accepting the setting of the directivity;
Based on the directivity setting accepted by the accepting means, the directivity setting means and a control means for instructing the adaptive filter to set directivity,
The adaptive filter reads a parameter corresponding to the setting of directivity instructed from the control unit from the storage unit, generates the pseudo regression sound signal using the read parameter, and then continues the previous sound collection signal. Generating a pseudo regression sound signal while updating the parameter based on the result of subtracting the pseudo regression sound signal at that time from
When an instruction for setting directivity not stored in the storage means is received, the pseudo-regressive sound signal is generated using predetermined parameters, and the pseudo-regressive sound signal at that time is continuously generated from the previous sound collection signal. When the pseudo-regressive sound signal is generated while the parameter is updated based on the subtraction result, and then an instruction for setting a new directivity is received, the undirected directivity setting and the time point used at that time are used. A recurring sound removal apparatus that additionally stores the stored parameters in the storage means. A speaker array comprising a plurality of speakers;
A microphone array consisting of multiple microphones;
Directivity setting means for setting directivity of the speaker array and the microphone array;
An adaptive filter that generates a pseudo-regression sound signal based on an audio signal input to the speaker array;
Regression sound removal means for subtracting the pseudo-regression sound signal from the sound collection signal output from the microphone array;
Storage means for storing the directivity setting and the adaptive filter parameter in association with each other;
Accepting means for accepting the setting of the directivity;
Based on the directivity setting accepted by the accepting means, the directivity setting means and a control means for instructing the adaptive filter to set directivity,
The adaptive filter reads a parameter corresponding to the setting of directivity instructed from the control unit from the storage unit, generates the pseudo regression sound signal using the read parameter, and then continues the previous sound collection signal. Generating a pseudo regression sound signal while updating the parameter based on the result of subtracting the pseudo regression sound signal at that time from
When an instruction for setting directivity not stored in the storage means is received, the pseudo-regressive sound signal is generated using predetermined parameters, and the pseudo-regressive sound signal at that time is continuously generated from the previous sound collection signal. When the pseudo-regressive sound signal is generated while the parameter is updated based on the subtraction result, and then an instruction for setting a new directivity is received, the undirected directivity setting and the time point used at that time are used. A recurring sound removal apparatus that additionally stores the stored parameters in the storage means. A plurality of channels of audio signals are input to the speaker array,
The directivity setting means sets directivity for each channel,
The adaptive filter and the regression sound removing means are provided for the number of the channels,
The regression sound removal apparatus according to claim 1 or 3, wherein the storage unit stores the directivity setting and parameters in association with each channel.

Images