Audio signal processing method, audio signal processing apparatus, audio signal processing system, and audio signal processing program

Abstract

This audio signal processing method used in an audio signal processing system provided with a microphone and a plurality of loudspeakers, said method comprising distributing an audio signal collected by the microphone to the plurality of loudspeakers, obtaining a loop gain to be fed back to the microphone for each of the plurality of loudspeakers, determining a target value of the loop gain for each of the plurality of loudspeakers, and setting a gain of the audio signal to be distributed to the plurality of loudspeakers, on the basis of the obtained loop gain and the target value of the loop gain.

Description

Audio signal processing method, audio signal processing apparatus, audio signal processing system, and audio signal processing program

[0001] One embodiment of the present invention relates to an audio signal processing method, an audio signal processing apparatus, an audio signal processing system, and an audio signal processing program.

[0002] Patent Document 1 discloses an audio signal processing method that changes to a minimum first transmission amount to be transmitted to a first sound emitting device closest to a first sound collecting device.

[0003] International Publication No. 2023 / 042699

[0004] The technique of Patent Document 1 minimizes the gain of the closest speaker, and does not consider loop gain.

[0005] An object of an embodiment of the present invention is to provide an audio signal processing method capable of providing amplified sound with better sound quality while ensuring amplified sound of sufficient sound pressure.

[0006] An audio signal processing method according to an embodiment of the present invention is an audio signal processing method used in an audio signal processing system including a microphone and a plurality of speakers, the method comprising: distributing an audio signal collected by the microphone to the plurality of speakers; obtaining, for each of the plurality of speakers, a loop gain fed back to the microphone; determining a target value of the loop gain for each of the plurality of speakers; and setting a gain of the audio signal to be distributed to the plurality of speakers based on the obtained loop gain and the target value of the loop gain.

[0007] The audio signal processing method according to an embodiment of the present invention can provide amplified sound with better sound quality while ensuring amplified sound of sufficient sound pressure.

[0008] This is a schematic block diagram showing the configuration of the sound signal processing system 1. This is a block diagram showing the configuration of the sound signal processing device 20. This is a diagram schematically showing the sound transmission path. This is a schematic block diagram schematically showing the mixing gain Gij in the sound signal processing device 20. This is a flowchart showing the operation of the sound signal processing device 20. This is a schematic block diagram showing the configuration of the sound signal processing system 1A according to modified example 1. This is a schematic block diagram schematically showing the equalizer gain Qij in the sound signal processing device 20. This is a diagram showing an example of the frequency characteristics of equalizer processing. This is a diagram showing an example of the frequency characteristics of equalizer processing when the distance between the speaker and the microphone is close and when it is far.

[0009] (First Embodiment) Figure 1 is a schematic block diagram showing the configuration of the sound signal processing system 1 according to the first embodiment. The sound signal processing system 1 comprises i microphones 10i (i=1 to m), a sound signal processing device 20, and j speakers 30j (j=1 to n).

[0010] The microphone 10i and speaker 30j are installed in the ceiling of the conference room 300, as an example. The microphone 10i and speaker 30j are connected to the sound signal processing device 20 by audio cables, communication cables, or wireless communication. For the sake of explanation, Figure 1 shows the sound signal processing device 20 as being installed in the ceiling, but in reality it is installed inside the conference room 300 or in a monitor room adjacent to the conference room 300. Furthermore, in this invention, the microphone 10i and speaker 30j do not need to be installed in the ceiling.

[0011] The sound signal processing system 1 is a system that amplifies the voice of speaker S and delivers it to listener L. Such a sound signal processing system 1 is used, for example, in meetings, seminars, or presentations. Figure 1 shows one speaker S and one listener L, but the number of speaker S and listener L is not limited to one each.

[0012] Microphone 10i captures the voice of speaker S. The sound signal processing device 20 receives the sound signal captured by microphone 10i. The sound signal processing device 20 performs signal processing such as mixing, gain adjustment, and equalization on the input sound signal. The sound signal processing device 20 outputs the processed sound signal to speaker 30j. Speaker 30j emits sound based on the input sound signal.

[0013] Figure 2 is a block diagram showing the configuration of the sound signal processing device 20. The sound signal processing device 20 consists of a general-purpose information processing device such as a personal computer or a smartphone, as an example. The general-purpose information processing device may be a device used by the speaker, or a device used by the operator running the conference or the installer setting up the conference system.

[0014] The sound signal processing device 20 includes a display 251, a user interface 252, a flash memory 253, a CPU 254, a RAM 255, and a communication interface 256.

[0015] The display unit 251 consists of, for example, an LCD or OLED, and displays various information. The user interface 252 is, for example, a touch panel stacked on the LCD or OLED of the display unit 251. Alternatively, the user interface 252 may be a keyboard or mouse. If the user interface 252 is a touch panel, the user interface 252, together with the display unit 251, constitutes a GUI (Graphical User Interface).

[0016] The communication interface 256 includes an audio interface and communication means such as wired LAN, wireless LAN, or Bluetooth®. The communication interface 256 is connected to the microphone 10i and the speaker 30j.

[0017] The CPU 254 is an example of a processor and is a control unit that controls the operation of the sound signal processing device 20. The CPU 254 performs various operations such as sound signal processing by reading a predetermined program, such as an application program, stored in the flash memory 253 (a storage medium) into the RAM 255 and executing it. The program may also be stored in a server (not shown). The CPU 254 may also download and execute a program from the server via a network.

[0018] Figure 3 is a schematic diagram showing the audio transmission path. Figure 4 is a schematic block diagram showing the gain of microphone 10i, the mixing gain in the sound signal processing device 20 (gain of the sound signal distributed to each speaker), and the gain of speaker 30j.

[0019] The speaker S's voice reaches the listener L via a sound amplification transmission path that passes through the microphone 10i, the sound signal processing device 20, and the speaker 30j. In addition, the speaker S's voice also reaches the listener L via a direct transmission path that passes through the space of the conference room 300 without passing through the sound amplification transmission path.

[0020] The transfer function of the direct transmission path from the speaker S to the listener L is H1, the transfer function from the speaker S to the microphone 10i is H2i, the gain of the microphone 10 is MGi, the gain of the sound signal distributed to each speaker for each microphone in the sound signal processing device 20 is Gij, the gain of speaker 30j is SGj, the transfer function from speaker 30j to the listening position is H3j, and the transfer function from speaker 30j to microphone 10i is H4ji.

[0021] The loop gain LGi of each microphone is expressed as the sum of the loop gains of each speaker, as shown in equation (1).

[0022] The transfer function from the speaker to the microphone increases as the speaker is closer to the microphone. On the other hand, the closer the speaker is to the microphone, the closer it is to the speaker's position, and the greater the sound pressure of the speaker's voice that reaches it through the direct transmission path. Therefore, the closer the speaker is to the microphone, the lower the speaker's gain can be.

[0023] Therefore, the sound signal processing device 20 of this embodiment determines the speaker gain Gij by setting a target value LG for the feedback (loop gain) from each speaker to the microphone. The target value LG for the loop gain from each speaker is made proportional to the distance d between the speaker and the microphone, as shown in equation (2), for example.

[0024] The coefficient w in equation (2) is a weighting coefficient and is not mandatory. The weighting coefficient w is a value of 0 or greater. When w = 0 in equation (2), the target value LG of the loop gain from each speaker is constant. However, even if the target value LG of the loop gain is constant, the speaker gain Gij increases the further it is from the microphone.

[0025] Figure 5 is a flowchart showing the operation of the sound signal processing device 20. First, the sound signal processing device 20 determines the maximum loop gain level (dB) for each speaker with respect to microphone i based on the distance between each microphone and each speaker, for example, as shown in the following formula (3) (S11).

[0026] l represents the speaker number. In this example, there is feedback from multiple speakers to a single microphone system, but the target loop gain from each speaker is set according to the distance, and the overall target loop gain for the microphone system is -6 dB. If the loop gain exceeds 0 dB, the feedback system may diverge and cause howling. Equation (3) means that the loop gain from all speakers to the microphone system is set to -6 dB, and the energy of the microphone system is distributed to each speaker with distance weighting. The coefficient w in equation (3) is a weighting coefficient and is not essential. The weighting coefficient w is a value of 0 or greater. The larger the value of the weighting coefficient w, the greater the gain of the speaker farther from the microphone.

[0027] The distance between each microphone and each speaker may be manually entered by the user, or it may be determined by means of imaging with a camera or measurement using LiDAR (Light Detection and Ranging). Alternatively, it may be determined by measuring the impulse response when a test sound is output from each speaker.

[0028] Subsequently, the sound signal processing device 20 determines the loop gain for each speaker (S12). The loop gain for each speaker is measured, for example, by outputting a test sound from speaker 30j. Then, the sound signal processing device 20 calculates the difference between the measured loop gain for each speaker and the maximum loop gain determined in S11 (S13). Based on the calculated difference, the sound signal processing device 20 sets the gain Gij for each speaker (S14). Note that the difference from the loop gain may be adjusted by the total gain of MGi, Gij, and SGj. Note that the gain Gij for each speaker for each microphone system may be determined by approximating the relationship between the value determined based on the loop gain and the distance between the microphone and the speaker using a curve. Specifically, the sound signal processing device 20 represents the relationship between distance and gain with a plurality of approximation curves and selects the approximation curve that is smallest at each distance. Then, the sound signal processing device 20 sets the gain according to the corresponding distance based on the selected approximation curve. This ensures that the same gain is set for speakers located at the same distance from each microphone, thereby suppressing differences in the amplification effect depending on the location and reducing discomfort for the listener. Furthermore, the approximation curve may be adjusted to the side where the gain is smallest, within the range where feedback does not occur. For example, by selecting the curve that gives the smallest gain from among several candidate curves, sufficient sound pressure can be ensured while prioritizing feedback suppression. The selection or correction of the approximation curve may also be performed by an operator via the user I / F 252. This operator may be used to adjust in the direction that reduces the gain difference due to distance. In addition, a lower limit and upper limit may be set for the value of Gij that is set. When the gain Gij is determined based on the loop gain, it may not be possible to set an appropriate value depending on the distance. For example, if Gij for a speaker very far from the microphone is determined based on the loop gain, the gain will be very large, and the amplified sound will be too loud. Therefore, the sound signal processing device 20 sets a lower limit when the maximum value of Gij is normalized to 0 dB. Also, the sound signal processing device 20 sets an upper limit when the maximum value of Gij is not normalized.

[0029] In this way, the sound signal processing device 20 determines the loop gain to be fed back to the microphone for each of the multiple speakers, determines a target value for the loop gain for each of the multiple speakers, and sets the gain Gij of the sound signal to be distributed to the multiple speakers based on the determined loop gain and the target value of the loop gain.

[0030] As a result of the above processing, as shown in Figure 3, the sound pressure of the speaker S's voice at the listener L's position is expressed as the sum of the amplification transmission path represented by gains H2i, MGi, Gij, SGj, and H3j, and the direct transmission path represented by gain H1. To suppress howling, the gain Gij is made smaller for speakers closer to the microphone, but the sound pressure of the direct transmission path increases. At positions farther from the speaker, the sound pressure of the direct transmission path decreases, but the gain Gij is made larger for speakers further from the microphone. Therefore, the sound signal processing device 20 can ensure sufficient sound pressure for amplified sound while suppressing howling, and can automatically provide amplified sound with better sound quality.

[0031] (Modification 1) Figure 6 is a schematic block diagram showing the configuration of the sound signal processing system 1A according to Modification 1. The microphone according to Modification 1 is an array microphone 100 in which a plurality of microphone units are arranged. The sound signal processing device 20 or the array microphone 100 sets up multiple sound pickup beams by combining the sound signals picked up by the plurality of microphone units. Beamforming is a delayed summing process that, for example, adds a delay to the sound signals of each microphone unit and combines them to form a sensitivity peak (focal point) at the speaker's position. i sound pickup beams (i = 1 to k) are set. In the example in Figure 6, three sound pickup beams b1, b2, and b3 are set.

[0032] The sound signal processing device 20 sets the gain Gij of the sound signal to be distributed to the multiple speakers for each of the multiple sound-collecting beams bi, rather than for each individual microphone system of the array microphone 100. That is, the gain Gij shown in the above embodiment corresponds to the gain for each of the multiple sound-collecting beams in Modification 1. For example, the sound signal processing device 20 sets the gain Gij of the sound signal to be distributed to the multiple speakers for each direction of the multiple sound-collecting beams. In this case, the sound signal processing device 20 may multiply the gain Gij by a correction value Gadjk (where k indicates the direction) for each direction.

[0033] In this case, the gain Gij increases with increasing speaker distance from the focal point of the sound-collecting beam and decreases with decreasing speaker distance. Therefore, the sound signal processing device 20 of Modification 1 can also ensure sufficient sound pressure for amplified sound while suppressing howling, and automatically provide amplified sound with better sound quality.

[0034] (Second Embodiment) The configuration of the sound signal processing system according to the second embodiment is the same as that shown in Figure 1, and the configuration of the sound signal processing device 20 is the same as that shown in Figure 2. In addition, the sound transmission path is the same as the transmission path shown in Figure 3.

[0035] Figure 7 is a schematic block diagram showing the gain Qij of the equalizer processing in the sound signal processing device 20 of the second embodiment. Figure 8 is a diagram showing an example of the frequency characteristics of the equalizer processing.

[0036] The sound signal processing device 20 according to the second embodiment performs equalizer processing to adjust the frequency characteristics of the sound signal to be distributed to the multiple speakers based on the distance between the microphone and each of the multiple speakers.

[0037] As shown in the frequency characteristics of Figure 8, the frequency characteristics of the equalizer processing in this example have the characteristic of suppressing the first frequency band (e.g., below 1 kHz) and emphasizing the second frequency band (e.g., the band above 1 kHz).

[0038] The directivity of speakers and microphones widens, especially in the low frequency range (e.g., below 1 kHz), and narrows in the high frequency range. Therefore, the loop gain from the speaker to the microphone increases in the low frequency range and decreases in the high frequency range. Thus, the sound signal processing device 20 can increase the amplification gain by increasing the amount of amplification in the high frequency range while suppressing howling by setting the frequency characteristics to further reduce the low frequency range. Such sound signal processing that changes the frequency characteristics (referred to as amplification gain equalizer processing) emphasizes the high frequencies of the sound signal in the amplification transmission path. However, if the sound in the amplification transmission path (referred to as amplified sound) is more dominant than the sound in the direct transmission path (referred to as direct transmission sound) among the sounds transmitted to the listener, such an emphasis on the high frequencies may be unpleasant to the listener.

[0039] Therefore, it is desirable to perform amplification that is suitable for the listener by changing the equalizer processing for amplification gain depending on the degree to which the amplified sound is dominant over the directly transmitted sound. In other words, if the directly transmitted sound is equal to or more dominant than the amplified sound among the sounds transmitted to the listener, equalizer processing should be applied to increase the amplification gain. Conversely, if the amplified sound is more dominant than the directly transmitted sound among the sounds transmitted to the listener, the degree of the equalizer for amplification gain should be reduced, or the equalizer for amplification gain should not be applied.

[0040] The sound signal processing device 20 sets the equalizer gain Qij inversely proportional to the distance d between the speaker and the microphone, as shown in equation (4), for example. Note that Qij is a gain with frequency characteristics.

[0041] The coefficient α in equation (4) is a weighting coefficient and is not mandatory. The weighting coefficient α is a value of 0 or greater. The larger the value of the weighting coefficient α, the smaller the correction amount for each frequency of the equalizer for the amplification gain of the speaker that is farther from the microphone.

[0042] As a result, the equalizer processing by the sound signal processing device 20 can increase the amplification gain at close range by emphasizing high frequencies that contribute to intelligibility compared to directly transmitted sound, while suppressing howling. On the other hand, at long distances, it is possible to transmit a natural amplified sound at a sufficient volume to the listener.

[0043] The frequency characteristic of the equalizer processing may be determined based on the sum of amplitude values over all frequencies of the transfer function between the speaker and the microphone, or the peak value thereof. That is, strong howling suppression equalizer processing is performed for speakers having a large loop gain that are prone to howling, and weak howling suppression equalizer processing is performed for speakers having a small loop gain.

[0044] This makes it possible to control equalizer processing more in accordance with the actual likelihood of howling occurrence.

[0045] The frequency characteristic of the equalizer processing may be determined stepwise according to a reference value x, such as the distance, or the sum of amplitude values over all frequencies of the transfer function, or the peak value. The following formula (6) is an example thereof. According to formula (6), the frequency characteristic becomes constant when the distance, or the sum of amplitude values over all frequencies of the transfer function, or the peak value is equal to or greater than the reference value.

[0046] Note that the first embodiment and the second embodiment can also be implemented in combination. When the first embodiment and the second embodiment are combined, the gain of the loudspeaker transmission path is represented by H2i·MGi·Qij·Gij·SGj·H3j, and the loop gain is represented by MGi·Qij·Gij·SGj·H4ji. In this case, the sound signal processing device 20 determines Gij after obtaining Qij. Specifically, the sound signal processing device 20 first obtains Qij by formulas (4) to (6) shown in the second embodiment. Then, the sound signal processing device 20 substitutes Qij into formulas (1) to (3), and obtains Gij using the obtained Qij.

[0047] (Modification 1 of the Second Embodiment) The frequency characteristic of the equalizer processing may be adjusted such that the closer the distance between the speaker and the microphone is, the more the first frequency band is suppressed and the second frequency band is emphasized. That is, the sound signal processing device 20 may determine the gain Qij of the equalizer processing based on the respective distances between the microphone and the plurality of speakers.

[0048] Figure 9 shows an example of the frequency characteristics of equalizer processing when the distance between the speaker and the microphone is short and when it is far. As shown in Figure 9, the sound signal processing device 20 suppresses low frequencies and emphasizes high frequencies more strongly when distributing sound signals to speakers at a short distance compared to when distributing signals at a long distance.

[0049] When the distance between the speaker and the microphone is short, the sound signal processing device 20 suppresses low frequencies more strongly to suppress howling, while strongly emphasizing high frequencies to increase the amplification gain. When the distance between the speaker and the microphone is far, the possibility of howling is low, and the sound pressure in the amplification transmission path is sufficiently greater than that in the direct transmission path, so the sound signal processing device 20 performs equalizer processing with a flat frequency response (or does not perform equalizer processing) to achieve a sound quality close to that of the direct transmission path. When equalizer processing is performed according to the distance between the microphone and the speaker, it is possible to provide an equalizer individually for all combinations of microphones and speakers, but this is not limited to this configuration. That is, the sound signal processing device 20 may classify the distance into multiple categories, for example, for short distance, medium distance, and long distance, pre-set equalizer characteristics corresponding to each category, and generate the output signal to each speaker through the equalizer of the corresponding category. By configuring it in this way, the sound signal processing device 20 can reduce the number of individual equalizers, which increases in proportion to the product of the number of microphones and the number of speakers, to a small number of equalizers corresponding to the number of distance divisions. In other words, the sound signal processing device 20 can consolidate the equalizer processing for input × output into multiple divisions such as for short distance, medium distance, and long distance, thereby reducing the computational load and design load. Furthermore, the sound signal processing device 20 can achieve frequency correction according to distance with a simple configuration by adjusting the gain of the output signal to the speakers belonging to each distance division.

[0050] Therefore, the sound signal processing device 20 of the second embodiment can also ensure sufficient sound pressure for amplified sound while suppressing howling, and provide amplified sound with better sound quality.

[0051] (Modification 2 of the Second Embodiment) Modification 2 of the second embodiment includes an array microphone 100 in which a plurality of microphone units are arranged, similar to Figure 6. The sound signal processing device 20 or the array microphone 100 sets up a plurality of sound-gathering beams by combining the sound signals picked up by the plurality of microphone units. Beamforming is a delayed summing process that, for example, adds a delay to the sound signals of each microphone unit and combines them to form a sensitivity peak (focal point) in the direction of the speaker. In the example of Figure 6, three sound-gathering beams b1, b2, and b3 are set.

[0052] The sound signal processing device 20 performs equalizer processing to adjust the frequency characteristics of the sound signals distributed to the multiple speakers, not for each individual microphone system of the array microphone 100, but for each of the multiple sound-collecting beams. For example, the gain Qij of the equalizer processing set for each of the i microphones 10i shown in the second embodiment above corresponds to the gain Qij of the equalizer processing for each of the multiple sound-collecting beams in the modified example 1. For example, the sound signal processing device 20 performs equalizer processing to adjust the frequency characteristics of the sound signals distributed to the multiple speakers for each direction of the multiple sound-collecting beams.

[0053] In this case, the gain Qij decreases for speakers further from the focal point of the sound-collecting beam and increases for speakers closer to the focal point. Therefore, the sound signal processing device 20 of the modified example 1 of the second embodiment can also ensure sufficient sound pressure for amplified sound while suppressing howling, and can automatically provide amplified sound with better sound quality.

[0054] (Modification 3 of the second embodiment) The sound signal processing device 20 of modification 3 of the second embodiment performs equalizer processing to adjust the frequency characteristics of the sound signal to be distributed to multiple speakers based on the type of microphone.

[0055] For example, the sound signal processing device 20 applies equalizer processing to a microphone installed on the ceiling, as shown in Figure 1, using the long-range frequency characteristics shown in Figure 9. On the other hand, the sound signal processing device 20 applies equalizer processing to a microphone installed on a table, using the short-range frequency characteristics shown in Figure 9.

[0056] A microphone mounted on the ceiling may not be able to capture the speaker's voice with sufficient sound pressure due to the distance from the speaker. Therefore, in the case of a microphone mounted on the ceiling, the sound signal processing device 20 suppresses low frequencies more strongly to suppress feedback, while strongly emphasizing high frequencies to increase the amplification gain. A microphone mounted on a tabletop has a low possibility of feedback due to the close distance from the speaker, and can capture the speaker's voice with sufficient sound pressure. Therefore, the sound signal processing device 20 performs equalizer processing with a flat frequency response (or does not perform equalizer processing) to achieve a sound quality close to that of a direct transmission path.

[0057] Therefore, the sound signal processing device 20 of the modified example 3 of the second embodiment can also ensure sufficient sound pressure for amplified sound while suppressing howling, and provide amplified sound with better sound quality.

[0058] The above description of embodiments should be considered in all respects to be illustrative and not restrictive. The scope of the present invention is indicated by the claims, not by the above embodiments. Furthermore, the scope of the present invention is intended to include all modifications within the meaning and scope equivalent to the claims.

[0059] 1, 1A: Sound signal processing system, 10i: Microphone, 20: Sound signal processing device, 30j: Speaker, 100: Array microphone, 251: Display, 252: User I / F, 253: Flash memory, 254: CPU, 255: RAM, 256: Communication I / F, 300: Conference room

Claims

1. A sound signal processing method used in a sound signal processing system comprising a microphone and a plurality of speakers, comprising: distributing a sound signal picked up by the microphone to the plurality of speakers; determining a loop gain to be fed back to the microphone for each of the plurality of speakers; determining a target value for the loop gain for each of the plurality of speakers; and setting the gain of the sound signal to be distributed to the plurality of speakers based on the determined loop gain and the target value for the loop gain.

2. The sound signal processing method according to claim 1, wherein the target value of the loop gain is determined based on the respective distances between the microphone and the plurality of speakers.

3. The sound signal processing method according to claim 1 or claim 2, further comprising equalizer processing to adjust the frequency characteristics of the sound signal distributed to the plurality of speakers based on the distance between the microphone and the plurality of speakers, or the transfer function between the microphone and the plurality of speakers.

4. The sound signal processing method according to claim 3, wherein the equalizer processing is adjusted to suppress the first frequency band and emphasize the second frequency band which is higher than the first frequency as the distance decreases.

5. The sound signal processing method according to claim 3, wherein the frequency characteristics are constant when the distance or the value of the transfer function is greater than or equal to a reference value.

6. The sound signal processing method according to claim 1 or 2, further comprising equalizer processing to adjust the frequency characteristics of the sound signal distributed to the plurality of speakers based on the type of microphone.

7. The sound signal processing method according to claim 1 or 2, wherein the microphone includes a plurality of microphones, and the gain of the sound signal to be distributed to each of the plurality of speakers is set for each of the plurality of microphones.

8. The sound signal processing method according to claim 1 or 2, wherein the microphone is an array microphone in which a plurality of microphone units are arranged, a plurality of sound pickup beams are set by combining sound signals picked up by the plurality of microphone units, and the gain of the sound signal to be distributed to each of the plurality of speakers is set for each of the plurality of sound pickup beams.

9. The sound signal processing method according to claim 1 or 2, wherein the microphone is an array microphone in which a plurality of microphone units are arranged, a plurality of sound pickup beams are set by combining sound signals picked up by the plurality of microphone units, and equalizer processing is performed for each of the plurality of sound pickup beams to adjust the frequency characteristics of the sound signals to be distributed to the plurality of speakers.

10. An audio signal processing device equipped with a processor that distributes an audio signal picked up by a microphone to a plurality of speakers, determines the loop gain to be fed back to the microphone for each of the plurality of speakers, determines a target value for the loop gain for each of the plurality of speakers, and sets the gain of the audio signal to be distributed to the plurality of speakers based on the determined loop gain and the target value for the loop gain.

11. A sound signal processing system comprising: a microphone; a plurality of speakers; a sound signal processing device that distributes the sound signal picked up by the microphone to the plurality of speakers; determines the loop gain to be fed back to the microphone for each of the plurality of speakers; determines a target value for the loop gain for each of the plurality of speakers; and sets the gain of the sound signal to be distributed to the plurality of speakers based on the determined loop gain and the target value for the loop gain.

12. An audio signal processing program that causes an audio signal processing device to perform the following processes: distribute an audio signal picked up by a microphone to a plurality of speakers; determine the loop gain to be fed back to the microphone for each of the plurality of speakers; determine a target value for the loop gain for each of the plurality of speakers; and set the gain of the audio signal to be distributed to the plurality of speakers based on the determined loop gain and the target value for the loop gain.

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