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

Abstract

This audio signal processing method is 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, performing equalizer processing for flattening frequency characteristics of a loop gain to be fed back from the plurality of loudspeakers to the microphone, and setting a gain of the equalizer processing according to an operation amount of an operator for receiving an operation from a user.

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 adjusts a volume within a reference value so that howling does not occur when volume adjustment is performed with an operation member for volume adjustment after automatic adjustment.

[0003] International Publication No. 2021 / 085174

[0004] The technique of Patent Document 1 does not change sound quality by a user's operation.

[0005] An object of one embodiment of the present invention is to provide an audio signal processing method that allows intuitive manual adjustment of sound quality with a small number of adjustment procedures after automatic adjustment.

[0006] An audio signal processing method according to one 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; performing equalization processing for flattening frequency characteristics of loop gain fed back from the plurality of speakers to the microphone; and setting a gain of the equalization processing according to an operation amount of an operation member that receives an operation from a user.

[0007] The audio signal processing method according to one embodiment of the present invention enables intuitive manual adjustment of sound quality with a small number of adjustment procedures after automatic adjustment.

[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 showing the mixing gain Gij in the sound signal processing device 20. This is a schematic block diagram showing the equalizer processing gain Qij in the sound signal processing device 20. This is a diagram showing an example of the frequency characteristics of the equalizer processing. This is an example of the GUI displayed on the display unit 251. This is an example of the GUI displayed on the display unit 251. This is an example of the GUI displayed on the display unit 251. This is a schematic block diagram showing the configuration of the sound signal processing system 1A according to modified example 2. This is a diagram showing an example of the frequency characteristics of the equalizer processing when the distance between the speaker and the microphone is close and when it is far.

[0009] Figure 1 is a schematic block diagram showing the configuration of the sound signal processing system 1 according to this 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 the equalizer processing in the sound signal processing device 20 is Qij, the gain of the 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 sets the gain Gij of the sound signal to be distributed to the multiple speakers based on the distance between the microphone and each of the multiple speakers. The gain Gij 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. As a result, the speaker gain Gij increases as it is farther from the microphone. Equation (2) above can also be used as an equation that takes into account the statistical reflection component. In this case as well, the gain Gij is proportional to the distance d between the speaker and the microphone.

[0025] 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.

[0026] Figure 5 is a schematic block diagram showing the gain Qij of the equalizer processing in the sound signal processing device 20. Figure 6 is a diagram showing an example of the frequency characteristics of the equalizer processing.

[0027] The sound signal processing device 20 performs equalizer processing to adjust the frequency characteristics of the sound signal distributed to the multiple speakers based on the distance between the microphone and each of the multiple speakers.

[0028] As shown in the frequency characteristics of Figure 6, the frequency characteristics of the equalizer processing in this example have the characteristic of suppressing the first frequency band (e.g., below 500 Hz) and emphasizing the second frequency band (e.g., the band above 500 Hz).

[0029] The directivity of speakers and microphones is particularly wide in the low frequency range (e.g., below 500 Hz) and narrows in the high frequency range. Therefore, the loop gain from the speaker to the microphone is high in the low frequency range and low in the high frequency range. The sound signal processing device 20 then performs equalizer processing to flatten the frequency characteristics of the loop gain fed back from the speaker to the microphone, for example by increasing the amount of high-frequency amplification while suppressing howling by setting the frequency characteristics to further reduce the low frequency range. Note that flattening does not necessarily mean a perfectly flat state.

[0030] This allows the sound signal processing device 20 to increase the amplification gain. Such sound signal processing that changes the frequency characteristics (referred to as amplification gain equalizer processing) often 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 amplification 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, then such an emphasis on high frequencies may be unpleasant to the listener's ears.

[0031] 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.

[0032] 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 (3), for example. Note that Qij is a gain with frequency characteristics.

[0033] The coefficient α in equation (3) 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.

[0034] 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.

[0035] The frequency response of the equalizer processing may be determined based on the sum of the amplitude values ​​or peak values ​​of the speaker and microphone transfer functions across all frequencies. In other words, for speakers with high loop gain that are prone to feedback, strong feedback suppression equalizer processing is applied, while for speakers with low loop gain, weaker feedback suppression equalizer processing is applied.

[0036] This allows for more precise control of the equalizer processing, tailored to the actual likelihood of feedback occurring.

[0037] The frequency response of the equalizer processing may be determined in steps according to a reference value x, such as distance, or the sum or peak value of the amplitude values ​​over all frequencies of the transfer function. Equation (5) below is one example. According to equation (5), the frequency response is constant when the distance, or the sum or peak value of the amplitude values ​​over all frequencies of the transfer function is greater than or equal to the reference value.

[0038] Furthermore, the sound signal processing device 20 of this embodiment allows the frequency characteristics of the equalizer processing to be changed by user operation. Figures 7 and 8 show examples of GUIs displayed on the display 251.

[0039] The sound signal processing device 20 displays a slider as the first operator, the frequency characteristics of the equalizer processing, and the frequency characteristics of the loop gain as a GUI. The slider is movable between "Sound Quality" and "Sound Volume". Figure 7 shows the frequency characteristics of the equalizer processing and the loop gain when the user moves the slider to "Sound Volume". When the user moves the slider to "Sound Volume", the frequency characteristics of the equalizer processing, which were automatically determined as described above, are set. That is, the sound signal processing device 20 sets the frequency characteristics to reduce the low frequencies and emphasize the high frequencies in most cases in order to flatten the loop gain.

[0040] On the other hand, Figure 8 shows the frequency characteristics of the equalizer processing and the loop gain when the user moves the slider to "Sound Quality". When the user moves the slider to "Sound Quality", a flatter frequency characteristic is set. That is, the sound signal processing device 20 automatically adjusts the logarithm of the gain of the equalizer processing, which is 10 log, according to equations (3) to (5). １０ (Qij^2) is further multiplied by a coefficient less than 1. The absolute value of this coefficient becomes smaller as the slider position approaches "Sound Quality". As a result, after automatic adjustment by the sound signal processing device 20, the user can obtain a new customer experience in which they can manually adjust the sound quality intuitively and with fewer adjustment steps.

[0041] Furthermore, when adjusting the gain of equalizer processing, the loop gain changes. Particularly when changing the setting from "Sound Volume" toward "Sound Quality", there is a risk that howling is more likely to occur. Therefore, when the equalizer is adjusted, it is preferable to adopt a configuration that additionally and automatically sets a separate gain for correcting the amount of change in the loop gain. Alternatively, it is preferable to adopt a configuration that multiplies a coefficient in a direction of decreasing the loop gain for a frequency at which the loop gain increases. This allows the user to obtain a new customer experience in which after automatic adjustment by the audio signal processing device 20, the user can intuitively manually adjust the sound quality with few adjustment procedures and safely against howling.

[0042] (Modification 1) In formula (2) of the above embodiment, an example (first setting) in which the gain Gij is proportional to the distance d between the speaker and the microphone has been described. In the first setting, the gain Gij is normalized by the distance between the microphone and the speaker, and the same gain is obtained for speakers at the same distance from the microphone.

[0043] However, since the loop gain is affected by the acoustic conditions in the room, it is not necessarily proportional to the distance between the speaker and the microphone. Further, the transfer characteristic from the speaker to the listening position does not necessarily match the loop gain. Even if the listening positions are at the same distance from the speaker, if the gain of the speaker assigned to each listening position and the transfer characteristic from the speaker to the listening position greatly differ, there is a risk that variation may occur, such as the amplified sound of the speaker through the public address transmission path varying greatly depending on the position.

[0044] Therefore, the audio signal processing device 20 may perform a second setting for setting the gain Gij of the audio signal distributed to each of the plurality of speakers based on the sound pressure of the sound directly arriving from the speaker to the listener and the total value of the sound pressures of the sounds arriving from the plurality of speakers to the listener, such that the loop gain LGi does not exceed a predetermined value.

[0045] The sound signal processing device 20 obtains, for each microphone, the loop gain fed back to the microphone for each of the plurality of speakers. The loop gain for each speaker is measured, for example, by outputting a test sound from the speaker 30j. The sound pressure of the sound of the speaker S at the position of the listener L is expressed as the sum of a sound amplification transmission path represented by the gain H2i·MGi·Gij·SGj·H3j and a direct transmission path represented by the gain H1.

[0046] The sound signal processing device 20 of Modification 1 can switch between the first setting and the second setting through a user operation. FIG. 9 is an example of a GUI displayed on the display 251. The sound signal processing device 20 displays a slider as a second operator on the GUI. The slider allows "Normalize" to be adjusted between 0 and 1 (0% to 100%). When "Normalize" is set to 100%, the gain Gij is normalized by the distance between the microphone and the speaker, and speakers at the same distance from the microphone have the same gain (first setting). When "Normalize" is set to 0%, the gain Gij is normalized by the transfer function between the microphone and the speaker, and the gain is set based on the target value of the loop gain distributed to each speaker (second setting). The gain Gij in the second setting is set such that the loop gain LGi does not exceed a predetermined value, and based on the distance attenuation of the sound that directly reaches the listener from the speaker and the transfer characteristic of the sound that reaches the listener from the microphone via the plurality of speakers. As described above, a speaker closer to the microphone produces a higher sound pressure of the speaker's sound arriving via the direct transmission path, while a speaker farther from the microphone produces a lower sound pressure of the speaker's sound arriving via the direct transmission path. Therefore, the gain Gij reflects the distance attenuation of sound from the position of the speaker to the position of the listener.

[0047] More specifically, the target value LG of the loop gain is proportional to the distance d between the speaker and the microphone, for example, as shown in equation (6).

[0048] The coefficient w in equation (6) is a weighting coefficient and is not mandatory. The weighting coefficient w is a value of 0 or greater. When w = 0 in equation (6), 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.

[0049] 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 equation (7).

[0050] 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 (7) 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 (7) 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.

[0051] Subsequently, the sound signal processing device 20 calculates the difference between the loop gain for each speaker and the maximum loop gain obtained above. Based on the calculated difference, the sound signal processing device 20 sets the gain Gij for each speaker. Note that the difference from the loop gain may be adjusted using the total gain of MGi, Gij, and SGj.

[0052] Thus, the sound signal processing device 20 of the modified example 1 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.

[0053] 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 provide amplified sound with better sound quality.

[0054] However, even in the second setting, if the distance between the microphone and speaker is the same, it is preferable to set the gain Gij to the smallest gain Gij within that distance. In this case, the sound signal processing device 20 multiplies the gain Gij by a coefficient corresponding to the value of 0 to 1 input via the GUI in Figure 9. Alternatively, the sound signal processing device 20 may determine a function such as the envelope between the distance between the microphone and speaker and the gain Gij, and set the lower limit of this envelope as the value of the gain Gij.

[0055] In the first modified example, after automatic adjustment by the sound signal processing device 20, the user can obtain a new customer experience in which they can manually adjust the sound quality intuitively and with fewer adjustment steps.

[0056] (Modification 2) Figure 10 is a schematic block diagram showing the configuration of the sound signal processing system 1A according to Modification 2. The microphone according to Modification 2 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 10, three sound pickup beams b1, b2, and b3 are set.

[0057] 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 2. 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.

[0058] 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 the modified example 2 can also ensure sufficient sound pressure for amplified sound while suppressing howling, and automatically provide amplified sound with better sound quality.

[0059] (Modification 3) The frequency characteristics of the equalizer processing may be adjusted so that the closer the distance between the speaker and the microphone, the more the first frequency band is suppressed and the more 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 multiple speakers.

[0060] Figure 11 shows an example of the frequency characteristics of equalizer processing when the distance between the speaker and microphone is short and when it is far. As shown in Figure 11, the sound signal processing device 20 suppresses low frequencies more strongly and emphasizes high frequencies more strongly when distributing sound signals to speakers at a short distance compared to when it is far away.

[0061] When the distance between the speaker and 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 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.

[0062] Therefore, the sound signal processing device 20 of the modified example 3 can also ensure sufficient sound pressure for amplified sound while suppressing howling, and provide amplified sound with better sound quality.

[0063] (Modification 4) Modification 4, like Modification 2 in Figure 10, includes an array microphone 100 in which multiple microphone units are arranged. The sound signal processing device 20 or the array microphone 100 sets up multiple sound-gathering beams by combining the sound signals picked up by the multiple microphone units. Beamforming is a delayed summing process that, for example, adds a delay to the sound signals from each microphone unit and combines them to form a sensitivity peak (focal point) in the direction of the speaker. In the example in Figure 10, three sound-gathering beams b1, b2, and b3 are set.

[0064] 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 equalizer processing gain Qij set for each of the i microphones 10i shown in the above embodiment corresponds to the equalizer processing gain Qij for each of the multiple sound-collecting beams in Modification 4. The sound signal processing device 20 performs equalizer processing to adjust the frequency characteristics of the sound signals distributed to the multiple speakers, for example, for each direction of the multiple sound-collecting beams.

[0065] 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 Modification 4 can also ensure sufficient sound pressure for amplified sound while suppressing howling, and can automatically provide amplified sound with better sound quality.

[0066] (Modification 5) The sound signal processing device 20 of Modification 5 performs equalizer processing to adjust the frequency characteristics of the sound signal to be distributed to multiple speakers based on the type of microphone.

[0067] 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 11. 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 11.

[0068] 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.

[0069] Therefore, the sound signal processing device 20 of Modification 5 can also ensure sufficient sound pressure amplified sound while suppressing howling, and provide amplified sound with better sound quality. In any of the above embodiments, for example, instead of performing equalizer processing as described above, the first setting and the second setting may be adjusted according to the amount of operation of the second operator from the user, as shown in Modification 1. In this case, this embodiment is a sound signal processing method used in a sound signal processing system equipped with a microphone and a plurality of speakers, and distributes the sound signal picked up by the microphone to the plurality of speakers, and adjusts according to the amount of operation of the operator: a first setting that sets the gain of the sound signal distributed to each of the plurality of speakers based on the distance between the microphone and the plurality of speakers, and a second setting that sets the gain of the sound signal distributed to each of the plurality of speakers based on the distance attenuation of sound that comes directly from the speaker to the listener and the transmission characteristics of sound that comes from the microphone to the listener via the plurality of speakers, so that the loop gain does not exceed a predetermined value. Furthermore, the microphone may be an array microphone in which multiple microphone units are arranged, and multiple sound pickup beams may be set by combining the sound signals picked up by the multiple microphone units, and the gain of the sound signal to be distributed to each of the multiple speakers may be set for each of the multiple sound pickup beams. As shown in Modifications 2 and 4, the sound signal processing device 20 adjusts the gain of the sound signal or the equalizer processing gain for each of the multiple sound pickup beams, but the difference in the gain of the sound signal or the equalizer processing gain for each sound pickup beam (i.e., each direction) can be increased or decreased by a third operator different from the first and second operators.

[0070] 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.

[0071] 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

A sound signal processing method used in a sound signal processing system equipped with a microphone and multiple speakers, The sound signal picked up by the microphone is distributed to the multiple speakers. Equalizer processing is performed to flatten the frequency characteristics of the loop gain that is fed back from the plurality of speakers to the microphone. The gain of the equalizer processing is set according to the amount of operation of the control element that receives input from the user. Audio signal processing method. 　 The frequency characteristics have the ability to suppress a first frequency band of the sound signal distributed to the plurality of speakers and to emphasize a second frequency band that is higher than the first frequency. The sound signal processing method according to claim 1. 　 The gain is set based on the distance between the microphone and the plurality of speakers. The sound signal processing method according to claim 1 or claim 2. 　 The operator includes a first operator for setting the gain of the equalizer processing and a second operator different from the first operator. A first setting that sets the gain of the sound signal to be distributed to each of the multiple speakers based on the distance between the microphone and each of the multiple speakers, A second setting that sets the gain of the sound signal distributed to each of the plurality of speakers, based on the distance attenuation of sound traveling directly from the speaker to the listener and the transmission characteristics of sound traveling from the microphone to the listener via the plurality of speakers, so that the loop gain does not exceed a predetermined value. The amount of the second operator is adjusted according to the amount of the second operator. The sound signal processing method according to claim 1 or claim 2. 　 The microphone includes multiple microphones, For each of the multiple microphones, the gain of the equalizer processing for the sound signal distributed to each of the multiple speakers is set. The sound signal processing method according to claim 1 or claim 2. 　 The aforementioned microphone is an array microphone in which multiple microphone units are arranged, By combining the sound signals picked up by the aforementioned multiple microphone units, multiple sound pickup beams are set. For each of the multiple sound-collecting beams, the gain of the sound signal to be distributed to each of the multiple speakers is set. The sound signal processing method according to claim 1 or claim 2. 　 An audio signal processing device used in an audio signal processing system that includes a microphone and multiple speakers, The sound signal picked up by the microphone is distributed to the multiple speakers. Equalizer processing is performed to flatten the frequency characteristics of the loop gain that is fed back from the plurality of speakers to the microphone. The gain of the equalizer processing is set according to the amount of operation of the control element that receives input from the user. A sound signal processing device equipped with a processor. Mike and, Multiple speakers, The sound signal picked up by the microphone is distributed to the multiple speakers. Equalizer processing is performed to flatten the frequency characteristics of the loop gain that is fed back from the plurality of speakers to the microphone. The gain of the equalizer processing is set according to the amount of operation of the control element that receives input from the user. Sound signal processing device, A sound signal processing system equipped with [specific features / features]. 　 The sound signal picked up by the microphone is distributed to multiple speakers. Equalizer processing is performed to flatten the frequency characteristics of the loop gain that is fed back from the plurality of speakers to the microphone. The gain of the equalizer processing is set according to the amount of operation of the control element that receives input from the user. A sound signal processing program that instructs a sound signal processing unit to perform the processing.

Images