Sound signal processing method and sound signal processing device

Abstract

To provide assistance for a desired sound field to electronic musical instruments, etc.SOLUTION: A sound field assistance system 1B inputs a sound signal in-line by means of sound sources 611B-613B in a room 62B that output sound signals of electronic pianos and electric guitars, etc., in-line. The sound signals inputted in-line are distributed after being mixed, and a mixing signal for initial reflected sounds and a mixing signal for reverberant sounds are obtained. Also, an impulse response for initial reflected sounds and an impulse response for reverberant sounds are obtained from the impulse response of a preliminarily measured prescribed space. Then, the impulse response for initial reflected sounds and the mixing signal for initial reflected sounds are convoluted, with the result reproduced by speakers 51A-51D installed on a wall surface. Likewise, the impulse response for reverberant sounds and the mixing signal for reverberant sounds are convoluted, with the result reproduced by speakers 61A-61F installed on a ceiling.EFFECT: It is possible to individually control the initial reflected sounds and reverberant sounds and suppress the blurring of sound images, thus realizing a richer expanse of sound image and space.SELECTED DRAWING: Figure 13

Description

[Technical field]

[0001] An embodiment of the present invention relates to a sound signal processing method and a sound signal processing device for processing an input sound signal. [Background technology]

[0002] In facilities such as concert halls, various genres of music are performed, and such facilities require various acoustic characteristics (for example, reverberation characteristics).

[0003] However, in order to physically change the reverberation characteristics within a hall, it is necessary to change the size of the acoustic space, for example by moving the ceiling, which requires extremely large-scale equipment.

[0004] Therefore, for example, a sound field control device as shown in Patent Document 1 processes sound picked up by a microphone with a Finite Impulse Response (FIR) filter to generate reverberant sound, and outputs the reverberant sound from speakers installed in the hall, thereby supporting the sound field. [Prior art documents] [Patent documents]

[0005] [Patent Document 1] Japanese Patent Application Publication No. 6-284493 Summary of the Invention [Problem to be solved by the invention]

[0006] However, when trying to support a sound field using sounds picked up by a microphone in electronic musical instruments and the like, it is sometimes impossible to achieve the desired sound field support.

[0007] SUMMARY OF THE PREFERRED EMBODIMENTS An object of one embodiment of the present invention is to provide a sound signal processing method and a sound signal processing device that realizes a desired sound field support for an electronic musical instrument or the like. [Means for solving the problem]

[0008] The sound signal processing method includes inputting a sound signal through a line, controlling the volume of the sound signal input through the line, and generating an early reflection sound control signal from the volume-controlled sound signal. Effect of the Invention

[0009] The sound signal processing method can realize a desired sound field support for an electronic musical instrument or the like. [Brief description of the drawings]

[0010] [Figure 1] FIG. 2 is a transparent perspective view that illustrates a space of the first embodiment. [Diagram 2] 1 is a block diagram showing the configuration of a sound field support system according to a first embodiment. [Diagram 3] 4 is a flowchart showing the operation of the sound signal processing device. [Figure 4] 13A is a schematic diagram showing an example of classification of sound types in the time waveform of an impulse response used for filter coefficients, and FIG. 13B is a schematic diagram showing the time waveform of filter coefficients set in an FIR filter 24A. [Diagram 5] FIG. 2 is a schematic diagram showing an impulse response set in an FIR filter 24A. [Figure 6] 13 is a plan view showing a schematic relationship between a space 620 and a chamber 62. FIG. [Figure 7] FIG. 1 is a block diagram showing a minimum configuration of a sound field support system. [Figure 8] FIG. 11 is a transparent perspective view that illustrates a space of the second embodiment. [Figure 9] FIG. 11 is a plan view showing a schematic view of a space according to a second embodiment. [Figure 10] FIG. 11 is a block diagram showing the configuration of a sound field support system according to a second embodiment. [Figure 11] 10 is a flowchart showing the operation of the sound signal processing device of the second embodiment. [Figure 12] FIG. 11 is a block diagram showing a minimum configuration of a sound field support system according to a second embodiment. [Figure 13] FIG. 11 is a transparent perspective view that illustrates a space of the third embodiment. [Figure 14] FIG. 1 is a block diagram showing a configuration of a sound field support system. [Figure 15] 11 is a flowchart showing the operation of the sound signal processing device of the third embodiment. [Figure 16] 2 is a block diagram showing a configuration of a sound signal processing unit. FIG. [Figure 17] 2 is a block diagram showing a configuration of a sound signal processing unit. FIG. [Figure 18] 2 is a block diagram showing a configuration of a sound signal processing unit. FIG. [Figure 19] 2 is a block diagram showing a configuration of a sound signal processing unit. FIG. DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS

[0011] [Embodiment 1] Fig. 1 is a see-through perspective view that typically shows a room 62 that constitutes a space. Fig. 2 is a block diagram showing the configuration of the sound field support system 1.

[0012] The room 62 forms a space with a roughly rectangular parallelepiped shape. The sound source 61 is located on a stage 60 at the front of the room 62. The rear of the room 62 corresponds to the audience seats where the listeners sit. The shape of the room 62 and the arrangement of the sound sources are not limited to the example in FIG. 1. The sound signal processing method and sound signal processing device of the present invention can provide a desired sound field in any shape of space, and can realize a richer sound image and spatial expansion than ever before.

[0013] The sound field supporting system 1 is equipped with directional microphone 11A, directional microphone 11B, directional microphone 11C, omnidirectional microphone 12A, omnidirectional microphone 12B, omnidirectional microphone 12C, speaker 51A, speaker 51B, speaker 51C, speaker 51D, speaker 61A, speaker 61B, speaker 61C, speaker 61D, speaker 61E, and speaker 61F within a room 62.

[0014] The speakers 61A, 61B, 61C, 61D, 61E, and 61F correspond to a first speaker that outputs a reverberation sound control signal, and the speakers 51A, 51B, 51C, and 51D correspond to a second speaker that outputs an early reflection sound control signal.

[0015] The number of directional microphones and the number of omnidirectional microphones shown in Fig. 1 are three each. However, the sound field support system 1 only needs to include at least one microphone. The number of speakers is not limited to the number shown in Fig. 1. The sound field support system 1 only needs to include at least one speaker.

[0016] Directional microphone 11A, directional microphone 11B, and directional microphone 11C mainly pick up the sound of sound source 61 on the stage.

[0017] Omnidirectional microphone 12A, omnidirectional microphone 12B, and omnidirectional microphone 12C are installed on the ceiling. Omnidirectional microphone 12A, omnidirectional microphone 12B, and omnidirectional microphone 12C pick up the entire sound in room 62, including the direct sound of sound source 61 and the reflected sound in room 62, etc.

[0018] Speaker 51A, speaker 51B, speaker 51C, and speaker 51D are installed on the wall surface of room 62. Speaker 61A, speaker 61B, speaker 61C, speaker 61D, speaker 61E, and speaker 61F are installed on the ceiling of room 62. However, in the present invention, the installation positions of the microphones and speakers are not limited to this example.

[0019] 2, the sound field supporting system 1 includes a sound signal processing unit 10 and a memory 31 in addition to the components shown in FIG. 1. The sound signal processing unit 10 is mainly composed of a CPU and a DSP (Digital Signal Processor). The sound signal processing unit 10 functionally includes a sound signal acquiring unit 21, a gain adjusting unit 22, a mixer 23, an FIR (Finite Impulse Response) filter 24A, an FIR filter 24B, a level setting unit 25A, a level setting unit 25B, a matrix mixer 26, a delay adjusting unit 28, an output unit 27, an impulse response acquiring unit 151, and a level balance adjusting unit 152. The sound signal processing unit 10 is an example of the sound signal processing device of the present invention.

[0020] The CPU constituting the sound signal processing unit 10 reads out an operation program stored in the memory 31 and controls each component. The CPU functionally configures an impulse response acquisition unit 151 and a level balance adjustment unit 152 by the operation program. It is not necessary for the operation program to be stored in the memory 31. The CPU may download the operation program each time, for example, from a server (not shown).

[0021] 3 is a flowchart showing the operation of the sound signal processing unit 10. First, the sound signal acquiring unit 21 acquires sound signals (S11). The sound signal acquiring unit 21 acquires sound signals from the directional microphone 11A, the directional microphone 11B, the directional microphone 11C, the omnidirectional microphone 12A, the omnidirectional microphone 12B, and the omnidirectional microphone 12C. When the sound signal acquiring unit 21 acquires an analog signal, it converts the analog signal into a digital signal and outputs it.

[0022] Gain adjustment unit 22 adjusts the gain of the sound signals acquired from directional microphone 11A, directional microphone 11B, directional microphone 11C, omnidirectional microphone 12A, omnidirectional microphone 12B, and omnidirectional microphone 12C through sound signal acquisition unit 21. Gain adjustment unit 22 sets the gain of a directional microphone located closer to sound source 61 to a high value, for example. Note that gain adjustment unit 22 is not an essential component in the first embodiment.

[0023] The mixer 23 mixes the sound signals acquired from the directional microphone 11A, the directional microphone 11B, and the directional microphone 11C. The mixer 23 also distributes the mixed sound signal to a plurality of signal processing systems. The mixer 23 outputs the distributed sound signal to the FIR filter 24A. The mixer 23 also mixes the sound signals acquired from the non-directional microphone 12A, the non-directional microphone 12B, and the non-directional microphone 12C. The mixer 23 outputs the mixed sound signal to the FIR filter 24B.

[0024] In the example of FIG. 2, the mixer 23 mixes the sound signals acquired from the directional microphone 11A, the directional microphone 11B, and the directional microphone 11C into four signal processing systems in accordance with the speakers 51A, 51B, 51C, and 51D. The mixer 23 also mixes the sound signals acquired from the non-directional microphone 12A, the non-directional microphone 12B, and the non-directional microphone 12C into four signal processing systems. This one signal processing system corresponds to the speakers 61A to 61F. Hereinafter, the four signal processing systems corresponding to the speakers 61A to 61F are referred to as the first system. The four signal processing systems corresponding to the speakers 51A, 51B, 51C, and 51D are referred to as the second system.

[0025] The number of signal processing systems is not limited to this example. Sound signals acquired from omnidirectional microphone 12A, omnidirectional microphone 12B, and omnidirectional microphone 12C may be distributed to six first systems according to speakers 61A, 61B, 61C, 61D, 61E, and 61F. Mixer 23 is not an essential component in the first embodiment.

[0026] The mixer 23 may have an EMR (Electronic Microphone Rotator) function. The EMR is a method of flattening the frequency characteristics of a feedback loop by changing the transfer function between a fixed microphone and a speaker over time. The EMR is a function of constantly switching the connection relationship between the microphone and the signal processing system. The mixer 23 outputs the sound signals acquired from the directional microphone 11A, the directional microphone 11B, and the directional microphone 11C to the FIR filter 24A in such a manner as to switch the output destination. Alternatively, the mixer 23 outputs the sound signals acquired from the omnidirectional microphone 12A, the omnidirectional microphone 12B, and the omnidirectional microphone 12C to the FIR filter 24B in such a manner as to switch the output destination. In this way, the mixer 23 can flatten the frequency characteristics of the acoustic feedback system from the speaker to the microphone in the room 62.

[0027] Next, the impulse response acquisition unit 151 sets the filter coefficients of the FIR filter 24A and the FIR filter 24B (S12).

[0028] Here, we will explain the impulse response data set in the filter coefficients. Fig. 4(A) is a schematic diagram showing an example of classification of sound types in the time waveform of the impulse response used in the filter coefficients, and Fig. 4(B) is a schematic diagram showing the time waveform of the filter coefficients set in the FIR filter 24A. Fig. 5(A) and Fig. 5(B) are schematic diagrams showing the time waveform of the filter coefficients set in the FIR filter 24B.

[0029] As shown in Fig. 4(A), the impulse response can be divided into a direct sound, an early reflection sound, and a reverberation sound, which are arranged on the time axis. The filter coefficients set in the FIR filter 24A are set by the part of the early reflection sound excluding the direct sound and the reverberation sound in the impulse response, as shown in Fig. 4(B). The filter coefficients set in the FIR filter 24B are set by the reverberation sound excluding the direct sound and the early reflection sound in the impulse response, as shown in Fig. 5(A). Note that the FIR filter 24B may be set by the early reflection sound and the reverberation sound excluding the direct sound in the impulse response, as shown in Fig. 5(B).

[0030] The impulse response data is stored in the memory 31. The impulse response acquisition unit 151 acquires the impulse response data from the memory 31. However, the impulse response data does not need to be stored in the memory 31. The impulse response acquisition unit 151 may download the impulse response data each time, for example, from a server (not shown).

[0031] The impulse response acquisition unit 151 may acquire data of an impulse response in which only the early reflection sound is extracted in advance, and set it in the FIR filter 24A. Alternatively, the impulse response acquisition unit 151 may acquire data of an impulse response including a direct sound, an early reflection sound, and a reverberation sound, and extract only the early reflection sound and set it in the FIR filter 24A. Similarly, when only the reverberation sound is used, the impulse response acquisition unit 151 may acquire data of an impulse response in which only the reverberation sound is extracted in advance, and set it in the FIR filter 24B. Alternatively, the impulse response acquisition unit 151 may acquire data of an impulse response including a direct sound, an early reflection sound, and a reverberation sound, and extract only the reverberation sound and set it in the FIR filter 24B.

[0032] Fig. 6 is a plan view showing a schematic relationship between a space 620 and a room 62. As shown in Fig. 6, impulse response data is measured in advance in a predetermined space 620, such as a concert hall or a church, in which a sound field is to be reproduced. For example, the impulse response data is measured by emitting a test sound (pulse sound) at the position of a sound source 61 and collecting the sound with a microphone.

[0033] The impulse response data may be acquired at any position in the space 620. However, it is preferable to measure the impulse response data of the early reflection sound using a directional microphone installed near the wall surface. The early reflection sound is a reflected sound with a clear arrival direction. Therefore, by measuring the impulse response data using a directional microphone installed near the wall surface, it is possible to precisely acquire reflected sound data of the target space. On the other hand, reverberation sound is a reflected sound with an uncertain arrival direction. Therefore, the impulse response data of the reverberation sound may be measured using a directional microphone installed near the wall surface, or may be measured using a non-directional microphone separate from the one used for measuring the early reflection sound.

[0034] The FIR filter 24A convolves different impulse response data into the four sound signals of the second system, which is the signal flow in the upper part of Fig. 2. If there are multiple signal processing systems, the FIR filter 24A and the FIR filter 24B may be provided for each signal processing system. For example, four FIR filters 24A may be provided.

[0035] When using directional microphones installed near the wall as described above, impulse response data is measured by a separate directional microphone for each signal processing system. For example, as shown in Fig. 6, for a signal processing system corresponding to speaker 51D installed on the right rear side facing the stage 60, impulse response data is measured by directional microphone 510D installed near the wall on the right rear side facing the stage 60.

[0036] The FIR filter 24A convolves the impulse response data with each sound signal of the second system (S13). The FIR filter 24B convolves the impulse response data with each sound signal of the first system, which is the signal flow in the lower part of FIG. 2 (S13).

[0037] The FIR filter 24A generates an early reflection sound control signal that reproduces the early reflection sound of a predetermined space by convolving the input sound signal with the impulse response data of the set early reflection sound. The FIR filter 24B generates a reverberation sound control signal that reproduces the reverberation sound of a predetermined space by convolving the input sound signal with the impulse response data of the set reverberation sound.

[0038] The level setting unit 25A adjusts the level of the early reflection sound control signal (S14), and the level setting unit 25B adjusts the level of the reverberation sound control signal (S14).

[0039] The level balance adjustment section 152 sets the level adjustment amounts of the level setting section 25A and the level setting section 25B.

[0040] The level balance adjustment unit 152 refers to the respective levels of the early reflection sound control signal and the reverberation sound control signal, and adjusts the level balance between them. For example, the level balance adjustment unit 152 adjusts the balance between the level of the last component in time of the early reflection sound control signal and the level of the first component in time of the reverberation sound control signal. Alternatively, the level balance adjustment unit 152 may adjust the balance between the power of multiple components in the latter part in time of the early reflection sound control signal and the power of components in the first part in time of the reverberation sound control signal. In this way, the level balance adjustment unit 152 can individually control the sounds of the early reflection sound control signal and the reverberation sound control signal, and control them to an appropriate balance according to the space to which they are applied.

[0041] Next, the matrix mixer 26 distributes the input sound signal to the output system for each speaker. The matrix mixer 26 distributes the reverberation sound control signal of the first system to each output system of the speakers 61A to 61F, and outputs it to the delay adjustment unit 28. Since the second system already corresponds to the output system, the matrix mixer 26 outputs the early reflection sound control signal of the second system to the delay adjustment unit 28 as it is.

[0042] The matrix mixer 26 may adjust the gain of each output system, adjust the frequency characteristics, and so on.

[0043] The delay adjustment unit 28 adjusts the delay time according to the distance between the sound source 61 and the multiple speakers (S15). For example, the delay adjustment unit 28 sets the delay time to be shorter for the multiple speakers in ascending order of the distance between the sound source 61 and the speaker. This allows the delay adjustment unit 28 to adjust the phase of the reverberation sound control signal and the early reflection sound control signal output from the multiple speakers according to the positions of the multiple speakers from the sound source 61.

[0044] The output unit 27 converts the early reflection sound control signal and the reverberation sound control signal output from the delay adjustment unit 28 into analog signals. The output unit 27 also amplifies the analog signals. The output unit 27 outputs the amplified analog signals to the corresponding speakers (S16).

[0045] With the above configuration, the sound signal processing unit 10 acquires a sound signal, acquires an impulse response, convolves the impulse response of the early reflection sound from the impulse response into the sound signal, and outputs the sound signal convolved with the impulse response of the early reflection sound as an early reflection sound control signal that has been subjected to processing different from that of the reverberation sound control signal. In this way, the sound signal processing unit 10 realizes a richer sound image and a wider space than ever before.

[0046] In the first embodiment, for example, the following configurations are also possible, and the following advantageous effects can be achieved in each configuration.

[0047] (1-1) One embodiment of the present invention is a signal processing method that acquires a sound signal, acquires an impulse response, and convolves an impulse response of an early reflection sound from the impulse response into the sound signal to generate an early reflection sound control signal.

[0048] 7 is a block diagram showing the configuration of a sound signal processing unit 10A corresponding to the above-mentioned signal processing method. The sound signal processing unit 10A includes a sound signal acquiring unit 21A that acquires a sound signal from a directional microphone 11A, an impulse response acquiring unit 151A that acquires an impulse response, and a processing unit 204A that convolutes an impulse response of an early reflection sound out of the impulse responses into the sound signal, and outputs the sound signal convolved with the impulse response of the early reflection sound to a speaker 51A as an early reflection sound control signal that has been subjected to processing different from that of the reverberation sound control signal.

[0049] The sound signal acquiring unit 21A has the same function as the sound signal acquiring unit 21 shown in Fig. 2. The impulse response acquiring unit 151A has the same function as the impulse response acquiring unit 151 in Fig. 2. The processing unit 204A has the functions of the FIR filter 24A and the output unit 27 in Fig. 2.

[0050] The sound signal processing unit 10A, like the sound signal processing unit 10 in FIG. 2, achieves a richer sound image and a wider space than ever before.

[0051] (1-2) The processing unit may convolve an impulse response of a reverberation sound among the impulse responses with the sound signal to generate a reverberation control signal that does not include direct sound, perform separate signal processing on the early reflection sound control signal and the reverberation control signal, and output the reverberation control signal to a first speaker (the above-mentioned first system speaker) and output the early reflection sound control signal to a second speaker (the above-mentioned second system speaker).

[0052] However, an actual room will have more speakers than the example shown in Fig. 1. Of the second speakers (speakers of the second system described above) that output the early reflection sound control signal, a speaker installed near the first speaker (speaker of the first system described above) may output a reverberation sound control signal. In other words, of the multiple speakers of the second system, a speaker installed near the speaker of the first system may output a reverberation sound control signal in addition to the early reflection sound control signal.

[0053] Conversely, among the first speakers (speakers of the first system described above), a speaker installed near the wall surface may output the early reflection sound control signal. In other words, among the multiple speakers of the first system, a speaker installed near the speaker of the second system may output the early reflection sound control signal in addition to the reverberation sound control signal.

[0054] This allows the early reflection sound control signal and the reverberation sound control signal to be adjusted with an appropriate energy balance.

[0055] (1-3) The first speaker may have a wide directivity, and the second speaker may have a narrow directivity.

[0056] As described above, the early reflection sound is a reflection sound with a clear arrival direction and contributes to the subjective impression. Therefore, it is effective to use a narrow directivity for the second speaker, and it is possible to improve the controllability of the early reflection sound in the target space.

[0057] On the other hand, reverberation is a reflected sound that does not have a fixed direction of arrival and contributes to the reverberation of the space. Therefore, it is effective to use a wide directivity for the first speaker, which can improve the controllability of the reverberation in the target space.

[0058] (1-4) It is preferable that the level per second speaker is higher than that per first speaker.

[0059] As mentioned above, the early reflection sound is reflected less times than the reverberation sound that is reflected multiple times in the space. Therefore, the energy of the early reflection sound is higher than the energy of the reverberation sound. Therefore, by increasing the level of each second speaker, it is possible to improve the subjective impression that the early reflection sound has, and to increase the controllability of the early reflection sound.

[0060] (1-5) It is preferable that the number of second speakers is smaller than the number of first speakers.

[0061] As described above, by reducing the number of second speakers, it is possible to suppress an increase in unnecessary diffuse sound energy. In other words, it is possible to suppress the early reflection sound output from the second speaker from diffusing and reverberating in the room, and to suppress the reverberation of the early reflection sound from reaching the listener.

[0062] (1-6) It is preferable that the first speaker is installed on the ceiling of the room, and the second speaker is installed on one side of the room.

[0063] By placing the second speaker on the side of the room, which is closer to the listener, it is easier to control how the early reflected sound reaches the listener, and the controllability of the early reflected sound can be improved. Also, by placing the first speaker on the ceiling of the room, it is possible to suppress the difference in reverberation depending on the listener's position.

[0064] (1-7) It is preferable that the processing unit adjusts the level balance between the early reflection sound control signal and the reverberation sound control signal.

[0065] The processing unit can adjust the level balance individually, thereby adjusting the sounds of the early reflection sound control signal and the reverberation sound control signal to an appropriate energy balance.

[0066] (1-8) It is preferable that the sound signal acquisition unit separately acquires a first sound signal for generating the reverberation sound control signal and a second sound signal for convolving the impulse response of the early reflection sound. The first sound signal is a sound signal corresponding to the above-mentioned first system (sound signals acquired from omnidirectional microphone 12A, omnidirectional microphone 12B, and omnidirectional microphone 12C), and the second sound signal is a sound signal corresponding to the above-mentioned second system (sound signals acquired from directional microphone 11A, directional microphone 11B, and directional microphone 11C).

[0067] Reverberation is easily affected by the reverberation in the room. Early reflection sounds are easily affected by the sound of the sound source. Therefore, it is preferable that the first sound signal picks up the entire sound in the room, and the second sound signal picks up the sound of the sound source with a high signal-to-noise ratio.

[0068] (1-9) It is preferable that the first sound signal is picked up by an omnidirectional microphone, and the second sound signal is picked up by a directional microphone.

[0069] As in the above, the first sound signal is preferably obtained by collecting the entire sound in the room using, for example, an omnidirectional microphone, and the second sound signal is preferably obtained by collecting the sound of the sound source with a high signal-to-noise ratio using, for example, a directional microphone.

[0070] (1-10) It is preferable that the directional microphone is closer to the sound source than the omnidirectional microphone.

[0071] As in the above, it is preferable that the second sound signal picks up the sound of the sound source with a high S / N ratio, and therefore it is preferable that the directional microphone is close to the sound source.

[0072] (1-11) It is preferable that the impulse response is acquired by using a directional microphone near a wall of a predetermined space.

[0073] The impulse response can be measured using a directional microphone placed near the wall, allowing the reflected sound from the target space to be obtained with greater accuracy.

[0074] [Embodiment 2] A sound field support system 1A of the second embodiment will be described with reference to Fig. 8, Fig. 9, Fig. 10, and Fig. 11. Fig. 8 is a see-through perspective view showing a schematic view of a space 620. Fig. 9 is a plan view showing the space 620. Fig. 10 is a block diagram showing the configuration of the sound field support system 1A.

[0075] 11 is a flowchart showing the operation of the sound signal processing device. In this example, it is assumed that a sound source 61 moves on a stage 60, or that a plurality of sound sources 61 are present on the stage 60. Note that the same components as those in the above-mentioned first embodiment are given the same reference numerals, and the description thereof will be omitted.

[0076] As shown in FIGS. 8 and 9, the sound field support system 1A includes a speaker 52A, a speaker 52B, a speaker 52C, a speaker 52D, a speaker 52E, a speaker 53A, a speaker 53B, a speaker 53C, a speaker 53D, and a speaker 53E.

[0077] In this example, as shown in Fig. 8 and Fig. 9, the speaker 52A, the speaker 52B, the speaker 52C, the speaker 52D, and the speaker 52E belong to the 2-1 speaker group 520 (to the left of the center as one faces the stage 60) that outputs the early reflection sound control signal of the 2-1 system. Also, in this example, the speaker 53A, the speaker 53B, the speaker 53C, the speaker 53D, and the speaker 53E belong to the 2-2 speaker group 530 (to the right of the center as one faces the stage 60) that outputs the early reflection sound control signal of the 2-2 system. The dashed line in Fig. 9 indicates the 2-1 speaker group 520, and the dashed line indicates the 2-2 speaker group 530.

[0078] In the following description, speaker 52A, speaker 52B, speaker 52C, speaker 52D, and speaker 52E of 2-1 speaker group 520 are collectively referred to as speakers of 2-1 speaker group 520. In the following description, speaker 53A, speaker 53B, speaker 53C, speaker 53D, and speaker 53E of 2-2 speaker group 530 are collectively referred to as speakers of 2-2 speaker group 530.

[0079] As shown in Figures 8 and 9, sound field support system 1A is equipped with directional microphone 13A, directional microphone 13B, directional microphone 13C, directional microphone 13D, directional microphone 14A, directional microphone 14B, directional microphone 14C and directional microphone 14D in room 62.

[0080] In this example, directional microphone 13A, directional microphone 13B, directional microphone 13C, and directional microphone 13D are installed on the ceiling lined up in the X1 direction (left-right direction) shown in Figures 8 and 9. Also, in this example, directional microphone 14A, directional microphone 14B, directional microphone 14C, and directional microphone 14D are arranged on the ceiling lined up in the X1 direction (left-right direction) shown in Figures 8 and 9. Also, directional microphone 14A, directional microphone 14B, directional microphone 14C, and directional microphone 14D are arranged rearward (toward the audience seats when looking at stage 60 from the side) from directional microphone 13A, directional microphone 13B, directional microphone 13C, and directional microphone 13D in the Y1 direction (front-back direction).

[0081] As shown in Fig. 9, the directional microphone 13A, the directional microphone 13C, the directional microphone 14A, and the directional microphone 14C correspond to the speakers of the 2-1 speaker group 520. That is, the early reflection sound control signal of the 2-1 system is generated based on the sound signals collected by the directional microphone 13A, the directional microphone 13C, the directional microphone 14A, and the directional microphone 14C. Also, the directional microphone 13B, the directional microphone 13D, the directional microphone 14B, and the directional microphone 14D correspond to the speakers of the 2-2 speaker group 530. That is, the early reflection sound control signal of the 2-2 system is generated based on the sound signals collected by the directional microphone 13B, the directional microphone 13D, the directional microphone 14B, and the directional microphone 14D.

[0082] In the following description, directional microphone 13A, directional microphone 13C, directional microphone 14A, and directional microphone 14C are collectively referred to as directional microphones corresponding to 2-1 speaker group 520. In the following description, directional microphone 13B, directional microphone 13D, directional microphone 14B, and directional microphone 14D are collectively referred to as directional microphones corresponding to 2-2 speaker group 530.

[0083] 10, a sound signal processing unit 10A of the sound field support system 1A is configured by removing the FIR filter 24B and the level setting unit 25B from the sound field support system 1 of the first embodiment. However, the second embodiment may also include the FIR filter 24B and the level setting unit 25B to generate a reverberation sound control signal. In that case, the reverberation sound control signal may be output to any one of the speakers 52A to 53E, or may be output from another speaker.

[0084] The sound signal acquiring unit 21 acquires sound signals from a directional microphone corresponding to the 2-1 speaker group 520 and a directional microphone corresponding to the 2-2 speaker group 530 (see FIG. 10).

[0085] The gain adjustment unit 22 adjusts the gain of the sound signals acquired from the directional microphone corresponding to the 2-1 speaker group 520 and the directional microphone corresponding to the 2-2 speaker group 530 (see FIG. 11, S101).

[0086] In this example, the gain adjustment unit 22 sets different gains for each of the directional microphones corresponding to the 2-1 speaker group 520 and each of the directional microphones corresponding to the 2-2 speaker group 530.

[0087] The gain adjustment unit 22 sets the gain of the sound signal to be higher for the directional microphones corresponding to the 2-1 speaker group 520 in ascending order of distance to a speaker of the 2-1 speaker group 520 (eg, speaker 52A) in the left-right direction.

[0088] In addition, the gain adjustment unit 22 sets the gain of the sound signal of the directional microphone corresponding to the 2-1 speaker group 520 on the front-to-back direction (left-to-right direction on the paper in Figure 9) when looking at the stage 60 from the side (right side on the paper in Figure 9) to be lower than the gain of the sound signal of the directional microphone on the side closer to the audience seats (left side on the paper in Figure 9).

[0089] Similarly to the above, the gain adjustment unit 22 sets the gain of the sound signal to be higher for the directional microphones corresponding to the 2-2 speaker group 530 in ascending order of distance to a speaker of the 2-2 speaker group 530 (for example, speaker 53A) in the left-right direction.

[0090] In addition, the gain adjustment unit 22 sets the gain of the sound signal of the directional microphone corresponding to the 2-2 speaker group 530 on the front-to-back direction (left-to-right direction on the paper in Figure 9) when looking at the stage 60 from the side (right side on the paper in Figure 9) to be lower than the gain of the sound signal of the directional microphone on the side closer to the audience seats (left side on the paper in Figure 9).

[0091] For example, gain adjustment unit 22 sets the gain of directional microphone 14A to 0 dB, the gain of directional microphone 13A to −1.5 dB, the gain of directional microphone 14C to −3.0 dB, and the gain of directional microphone 13C to −4.5 dB.

[0092] For example, gain adjustment unit 22 sets the gain of directional microphone 14D to 0 dB, the gain of directional microphone 13D to −1.5 dB, the gain of directional microphone 14B to −3.0 dB, and the gain of directional microphone 13B to −4.5 dB.

[0093] The mixer 23 mixes the sound signals acquired from the directional microphones corresponding to the 2-1 speaker group 520 (see S102 in FIG. 11). The mixer 23 distributes the mixed sound signals to a number of signal processing systems (five in FIG. 8 and FIG. 9) according to the number (e.g., five) of speakers in the 2-1 speaker group 520. The mixer 23 also mixes the sound signals acquired from the directional microphones corresponding to the 2-2 speaker group 530. The mixer 23 distributes the mixed sound signals to a number of signal processing systems (five in FIG. 8 and FIG. 9) according to the number (e.g., five) of speakers in the 2-2 speaker group 530.

[0094] In a real space, sound image localization changes depending on the direction of arrival, level, and density of the reflected sound of the direct sound and the early reflected sound. That is, the sound image localization of the sound source 61 in the audience seats depends on the position of the sound source 61 on the stage 60. For example, when the sound source 61 moves to the left side toward the stage 60, the direct sound and early reflected sound levels arriving from the left direction become relatively high in the audience seats, so that the sound image is localized to the left side toward the stage 60. The gain adjustment unit 22 sets the gain of the sound signal to be high for the multiple directional microphones in order of the distance to the speaker, thereby controlling the level of the early reflected sound according to the position of the sound source 61 on the stage 60, thereby realizing sound image localization close to the phenomenon in the real space.

[0095] The delay adjustment unit 28 adjusts the delay time according to the distance between the multiple directional microphones and the speaker. For example, the delay adjustment unit 28 sets the delay time to be shorter for the multiple directional microphones in ascending order of the distance between the directional microphone and the speaker. As a result, the time difference between the early reflection sounds output by the multiple speakers is reproduced according to the distance between the sound source 61 and the speaker.

[0096] Moreover, the sound field supporting system 1A acquires the sound of the sound source 61 over a wide range on the stage 60 by arranging multiple directional microphones in the left-right direction. This allows the sound field supporting system 1A to reflect the level of the early reflected sound according to the position of the sound source 61 in a state close to the real space, without detecting the position of the sound source 61.

[0097] In addition, in the real space, as the distance between the sound source 61 and the audience seats increases, the level of the early reflection sound also decreases. The gain adjustment unit 22 realizes the reverberation of the sound in the real space by setting a low gain for the sound signal of the speaker that is farther from the audience seats in the front-to-rear direction.

[0098] Furthermore, in the real space, as the distance between the sound source 61 and the audience seats increases, the time it takes for the direct sound to reach the audience seats from the sound source 61 increases. Therefore, by using the delay adjustment unit 28 to set a large delay time for the early reflection sound signal output to the speaker farther from the audience seats, the sound field support system 1A realizes the reverberation of sound in the real space more accurately.

[0099] In this way, even when the sound source 61 moves on the stage 60 or there are multiple sound sources 61, the sound field support system 1A of the second embodiment can generate an early reflection sound control signal corresponding to the position of the sound source 61 without separately acquiring position information of the sound source 61 by setting the gain of the directional microphone according to the positional relationship between the sound source and the speaker. Therefore, the sound field support system 1 can effectively realize sound image localization, and can realize a richer sound image and a wider space than ever before.

[0100] The gain value of the sound signal of the directional microphone is not limited to this example. Also, although the example has been described in which the gain of the sound signal of the speaker farther from the audience seats is set lower than the gain of the sound signal of the speaker closer to the audience seats, the present invention is not limited to this example.

[0101] In addition, in the sound field support system 1A of the second embodiment, eight directional microphones are used, but the present invention is not limited to this. The number of directional microphones may be less than eight or may be nine or more. The positions of the directional microphones are also not limited to this example.

[0102] In addition, in the sound field support system 1A of the second embodiment, five speakers in the 2-1 speaker group 520 and five speakers in the 2-2 speaker group 530 have been described, but this is not limited to this. The number of speaker groups may be three or more, and the number of speakers belonging to each speaker group may be one or more. Furthermore, the positions of the speakers are not limited to this example.

[0103] Furthermore, in the sound field support system 1A of the second embodiment, one directional microphone may be associated with, for example, both the 2-1 speaker group 520 and the 2-2 speaker group 530. In this case, the gain of the sound signal corresponding to the 2-1 speaker group 520 (2-1 system) may be different from the gain of the sound signal corresponding to the 2-2 speaker group 530 (2-2 system).

[0104] In the second embodiment, for example, the following configurations are also possible, and the following advantageous effects can be achieved in each configuration.

[0105] (2-1) A sound signal processing method acquires a plurality of sound signals picked up by a plurality of microphones arranged in a specified space, adjusts the levels of the plurality of sound signals according to the respective positions of the plurality of microphones, mixes the adjusted plurality of sound signals, and generates a reflected sound using the mixed signal.

[0106] 12 is a block diagram showing the configuration of a sound signal processing unit 10C corresponding to the signal processing method of embodiment 2. The sound signal processing unit 10C includes a sound signal acquisition unit 21B that acquires a plurality of sound signals picked up by a plurality of directional microphones 13A, 13B, 14A, and 14B arranged in a predetermined space, a gain adjustment unit 22B that adjusts the levels of the plurality of sound signals according to the respective arrangement positions of the plurality of directional microphones 13A, 13B, 14A, and 14B, a mixer 23B that mixes the adjusted plurality of sound signals, and a reflected sound generation unit 205B that uses the mixed mixed signals to generate reflected sounds for each system and output the reflected sounds to the speakers 52A and 53A.

[0107] The sound signal acquiring unit 21B has the same function as the sound signal acquiring unit 21 shown in Fig. 10. The gain adjusting unit 22B has the same function as the gain adjusting unit 22 in Fig. 10. The mixer 23B has the same function as the mixer 23 in Fig. 10. The reflected sound generating unit 205B has the same function as the FIR filter 24A and the level setting unit 25A in Fig. 10.

[0108] Similar to the sound signal processing unit 10B in FIG. 10, the sound signal processing unit 10C achieves more effective sound image localization by changing the level of the signal collected from the sound signal acquisition unit 21B according to the position of the sound source, without the need to detect the position of the sound source.

[0109] (2-2) The levels of the respective sound signals may be adjusted in accordance with the distance from an arrangement position of each of the plurality of microphones to a speaker that outputs the reflected sound.

[0110] In a real space, the positioning of the sound image changes depending on the direction of arrival of the direct sound and early reflected sound, as well as the level and density of the reflected sound. Therefore, this configuration reproduces the reverberation of sound in a real space more accurately.

[0111] (2-3) In the level adjustment, a gain for each of the plurality of sound signals may be set to be higher in an order of decreasing distance from the position of each of the plurality of microphones to the position of the speaker that outputs the reflected sound.

[0112] In this configuration, the gain of the sound signal is set higher for the directional microphones closest to the speaker, reproducing the attenuation of reflected sound that depends on the distance between the sound source and the wall, and further realizing the reverberation of sound in a real space.

[0113] (2-4) A delay adjustment may be performed according to the distance from each of the positions of the microphones to a speaker that outputs the reflected sound. With this configuration, sound image localization close to a phenomenon in a real space is realized.

[0114] (2-5) The delay time may be set to be longer as the distance from each of the positions of the plurality of microphones to the speaker that outputs the pseudo-reflected sound increases.

[0115] This configuration reproduces the delay of reflected sound that depends on the distance between the sound source and the wall.

[0116] (2-6) The sound signal generating device may include speakers that output reflected sounds, the speakers that output the reflected sounds may include a 2-1 speaker group of a 2-1 system and a 2-2 speaker group of a 2-2 system, the level adjustment unit may adjust the level of each sound signal for each of the 2-1 system and the 2-1 system, and the mixing unit may mix each of the 2-1 system and the 2-2 system.

[0117] With this configuration, sound image localization can be achieved more effectively.

[0118] (2-7) The sound signal generating device includes a plurality of microphones arranged in a predetermined space, and it is preferable that the plurality of microphones are divided into a plurality of 2-1 microphones corresponding to the 2-1 speaker group and a plurality of 2-2 microphones corresponding to the 2-2 speaker group.

[0119] With this configuration, even if the position of the sound source moves or there are multiple sound sources, it is possible to more effectively achieve sound image localization.

[0120] (2-8) The reflected sound may include an early reflected sound.

[0121] [Embodiment 3] The sound field support system 1B of the third embodiment will be described with reference to Figs. 13, 14, and 15. Fig. 13 is a perspective view showing a schematic view of the room 62B of the third embodiment. Fig. 14 is a block diagram showing the configuration of the sound field support system 1B. Fig. 15 is a flow chart showing the operation of the sound signal processing device of the third embodiment. The third embodiment assumes that the output sounds from the sound source 611B, the sound source 612B, and the sound source 613B are inputted as line input. The same components as those of the first embodiment are given the same reference numerals and will not be described. The line input is not the input of the sound outputted from the sound source such as various musical instruments described later by a microphone, but the input of the sound signal from an audio cable connected to the sound source. In contrast, the line output is the input of the sound signal from the sound source such as various musical instruments described later by an audio cable connected to the sound source. The room 62B does not require the directional microphone 11A, the directional microphone 11B, and the directional microphone 11C, as compared with the room 62 shown in the first embodiment. Note that directional microphone 11A, directional microphone 11B, and directional microphone 11C may be arranged.

[0122] Sound source 611B, sound source 612B, and sound source 613B are, for example, an electronic piano, an electric guitar, etc., and each outputs a sound signal via a line. That is, sound source 611B, sound source 612B, and sound source 613B are connected to an audio cable, and output a sound signal via the audio cable. Note that, although the number of sound sources is three in FIG. 13, it may be one, or it may be two or more than four.

[0123] A sound signal processing unit 10D of the sound field support system 1B differs from the sound signal processing unit 10 shown in the first embodiment in that it further includes a line input unit 21D, a sound signal acquiring unit 210, a level setting unit 211, a level setting unit 212, a synthesis unit 213, and a mixer 230. Other configurations of the sound signal processing unit 10D are similar to those of the sound signal processing unit 10, and a description of similar parts will be omitted.

[0124] The line input unit 21D inputs audio signals from the sound source 611B, the sound source 612B, and the sound source 613B (see S201 in FIG. 15). That is, the line input unit 21D is connected to an audio cable connected to the sound source 611B, the sound source 612B, and the sound source 613B. The line input unit 21D then inputs the audio signals from the sound source 611B, the sound source 612B, and the sound source 613B via this audio cable. Hereinafter, these audio signals are referred to as line input signals. The line input unit 21D outputs the line input signals of the respective sound sources to the gain adjustment unit 22.

[0125] The gain adjustment unit 22 corresponds to a volume control unit, and performs volume control of the line input signal (see S202 in FIG. 15). Specifically, the gain adjustment unit 22 performs volume control using individual gains for the line input signal of the sound source 611B, the line input signal of the sound source 612B, and the line input signal of the sound source 613B. The gain adjustment unit 22 outputs the line input signal after the volume control to the mixer 23.

[0126] The mixer 23 mixes the volume-controlled line input signal of the sound source 611B, the volume-controlled line input signal of the sound source 612B, and the volume-controlled line input signal of the sound source 613B.

[0127] The mixer 23 distributes the mixed sound signal to a plurality of signal processing systems. Specifically, the mixer 23 distributes the mixed sound signal to a plurality of signal processing systems for early reflection sounds and a signal processing system for reverberation sounds. Hereinafter, the sound signal distributed to the plurality of signal processing systems for early reflection sounds is referred to as a mixed signal for early reflection sounds, and the sound signal distributed to the signal processing system for reverberation sounds is referred to as a mixed signal for reverberation sounds.

[0128] The mixer 23 outputs a mixed signal for the early reflection sound to the level setting section 211. The mixer 23 outputs a mixed signal for the reverberation sound to the level setting section 212.

[0129] The level setting unit 211 adjusts the level of a mixing signal for early reflection sounds. The level setting unit 212 adjusts the level of a mixing signal for reverberation sounds. The level adjustments of the level setting units 211 and 212 are set by the level balance adjustment unit 152, similar to the level setting units 25A and 25B.

[0130] The level setting unit 211 outputs the level-adjusted mixing signal for the early reflection sound to the FIR filter 24 A. The level setting unit 212 outputs the level-adjusted mixing signal for the reverberation sound to the synthesis unit 213.

[0131] The sound signal acquiring unit 210 acquires picked-up signals from the omnidirectional microphone 12A, the omnidirectional microphone 12B, and the omnidirectional microphone 12C. The sound signal acquiring unit 210 outputs the acquired picked-up signals to the mixer 230. The mixer 230 mixes the picked-up signals from the sound signal acquiring unit 210. The mixer 230 outputs the mixed picked-up signals to the synthesis unit 213.

[0132] The synthesis unit 213 synthesizes (adds) the level-adjusted mixed signal for reverberation sound from the level setting unit 212 and the mixed picked-up signal from the mixer 230. The synthesis unit 213 outputs the synthesized signal to the FIR filter 24B.

[0133] The FIR filter 24A convolves the level-adjusted mixing signal for early reflection sounds with an impulse response for early reflection sounds to generate an early reflection sound control signal. The FIR filter 24B convolves the synthesis signal with an impulse response for reverberation sounds to generate a reverberation sound control signal.

[0134] The level setting unit 25A adjusts the level of the early reflection sound control signal, and the level setting unit 25B adjusts the level of the reverberation sound control signal.

[0135] The matrix mixer 26 distributes the input sound signal to the output system for each speaker. The matrix mixer 26 distributes the reverberation sound control signal to each output system of the speakers 61A to 61F and outputs it to the delay adjustment unit 28. The matrix mixer 26 distributes the early reflection sound control signal to each output system of the speakers 51A to 51D and outputs it to the delay adjustment unit 28.

[0136] The delay adjustment unit 28 adjusts the delay time according to the distance between the sound source 611B, the sound source 612B, and the sound source 613B and the multiple speakers. This allows the delay adjustment unit 28 to adjust the phase of the reverberation sound control signal and the early reflection sound control signal output from the multiple speakers according to the positional relationship (distance) between the sound source 611B, the sound source 612B, and the sound source 613B and the multiple speakers.

[0137] The output unit 27 converts the early reflection sound control signal and the reverberation sound control signal output from the delay adjustment unit 28 into analog signals. The output unit 27 also amplifies the analog signals. The output unit 27 outputs the amplified analog signals to the corresponding speakers.

[0138] By using this configuration and performing this processing, the sound signal processor 10D can realize a richer sound image and a wider space than ever before for a line input signal (a sound signal input through a line). Therefore, the sound signal processor 10D can realize a desired sound field support for a sound source having a line output such as an electronic musical instrument.

[0139] Furthermore, the sound signal processing unit 10D generates an early reflection sound control signal using a line input signal. The line input signal has a higher S / N ratio than a sound signal picked up by a microphone. Therefore, the sound signal processing unit 10D can generate the early reflection sound control signal without being affected by noise. This allows the sound signal processing unit 10D to more reliably realize a desired sound field having a richer sound image and a wider spatial spread than ever before.

[0140] Furthermore, the sound signal processing unit 10D controls the volume of the line input signal, and generates the early reflection sound control signal using the line input signal after volume control. Each electronic musical instrument has a different default volume level. Therefore, if volume control is not performed, for example, when the electronic musical instrument that is line-input is switched, the desired early reflection sound control signal cannot be generated. However, by controlling the volume of the line input signal, the sound signal processing unit 10D can keep the level of the sound signal used to generate the early reflection sound control signal constant. As a result, the sound signal processing unit 10D can generate the desired early reflection sound control signal even when, for example, the electronic device that is line-input is switched.

[0141] Furthermore, the sound signal processing unit 10D mixes a plurality of line input signals after controlling their volumes. Then, the sound signal processing unit 10D generates an early reflection sound control signal using the mixed sound signal. This allows the sound signal processing unit 10D to properly adjust the level balance of the plurality of line input signals. Therefore, even if there are a plurality of line input signals, the sound signal processing unit 10D can generate a desired early reflection control signal.

[0142] The sound signal processor 10D can obtain these operational effects not only on the early reflection sound control signal but also on the reverberation sound control signal.

[0143] Moreover, the sound signal processing unit 10D uses only the line input signal to generate the early reflection sound control signal. On the other hand, the sound signal processing unit 10D uses the line input signal and the sound signal collected by the omnidirectional microphone to generate the reverberation sound control signal. By individually controlling the early reflection sound and the reverberation sound, the blurring of the sound image is suppressed, and a rich sound image and spatial expansion are realized. Furthermore, by using the sound signal collected by the omnidirectional microphone for the reverberation sound control signal, the effect of the sound field support can be extended not only to the sound of a sound source such as an electronic musical instrument, but also to sounds generated within a space such as the applause of an audience. Therefore, by having this configuration, the sound signal processing unit 10D can realize flexible sound field support.

[0144] Although the above description does not include a description of reproduction of direct sound, the sound signal processor 10D may include a direct sound processing system as a processing system separate from the above-described configuration.

[0145] In this case, for example, the sound signal processing unit 10D performs level adjustment on the output of the mixer 23, that is, the mixed sound signal, and outputs the signal to separately installed stereo speakers or the like.

[0146] Furthermore, for example, the sound signal processing unit 10D performs level adjustment on the mixed sound signal and outputs it to the matrix mixer 26. The matrix mixer 26 mixes the direct sound signal, the early reflection sound control signal, and the reverberation sound control signal, and outputs it to the output unit 27. At this time, the matrix mixer 26 may mix the direct sound signal, the early reflection sound control signal, and the reverberation sound control signal so as to set a dedicated speaker for the direct sound signal and output the direct sound signal to this dedicated speaker.

[0147] In the above description, the sound source 611B, the sound source 612B, and the sound source 613B are electronic musical instruments as an example. However, the sound source 611B, the sound source 612B, and the sound source 613B may be a hand microphone held by the singer, a stand microphone installed near the singer, or the like, which is placed near the singer, picks up the voice of the singer, and outputs a singing sound signal.

[0148] In the third embodiment, for example, the following configurations are also possible, and the following advantageous effects can be achieved in each configuration. Note that in the following description, the same parts as those described above will not be described.

[0149] (3-1) One embodiment of the third embodiment of the present invention is a sound signal processing method for inputting a sound signal through a line, controlling the volume of the line-input sound signal, and generating an early reflection sound control signal from the volume-controlled sound signal.

[0150] 16 is a block diagram showing the configuration of a sound signal processing unit 10E corresponding to the above-mentioned sound signal processing method. The sound signal processing unit 10E includes a line input unit 21E, a gain adjustment unit 22E, an early reflection sound control signal generation unit 214, an impulse response acquisition unit 151A, and a delay adjustment unit .

[0151] The line input unit 21E receives one line input signal and outputs it to the gain adjustment unit 22E. The gain adjustment unit 22E controls the volume of the line input signal. The gain adjustment unit 22E outputs the volume-controlled line input signal to the early reflection sound control signal generation unit 214.

[0152] The early reflection sound control signal generating unit 214 convolves the volume-controlled line input signal with impulse response data for the early reflection sound to generate an early reflection sound control signal. As in the above-described embodiment, the early reflection sound control signal generating unit 214 obtains impulse response data from a memory, for example, and uses the data for the convolution. The early reflection sound control signal generating unit 214 outputs the early reflection sound control signal to the delay adjustment unit 28. As in the above-described description, the delay adjustment unit 28 adjusts the delay time of the early reflection sound control signal and outputs it to the speaker 51A. Note that, when there are multiple speakers, a matrix mixer 26 may be provided similarly to the above-described sound signal processing unit 10. The matrix mixer 26 distributes the early reflection sound control signal to multiple speakers and outputs it.

[0153] With this configuration and method, the sound signal processing unit 10E can appropriately generate an early reflection sound control signal for one line input signal, and can realize a desired sound field having a richer sound image and a wider spatial spread than ever before.

[0154] (3-2) One embodiment of the third embodiment of the present invention is a sound signal processing method in which there are a plurality of line inputs, and sound signals input through a plurality of lines are subjected to volume control for each line input.

[0155] With this configuration and method, the sound signal processing unit can appropriately generate an early reflection sound control signal for a plurality of line input signals, and can realize a desired sound field having a richer sound image and a wider space than ever before. Also, the sound signal processing unit can appropriately adjust the level balance between a plurality of line input signals, and can realize a desired sound field having a richer sound image and a wider space.

[0156] (3-3) One embodiment of the third embodiment of the present invention is a sound signal processing method for mixing a plurality of line-input sound signals and generating an early reflection sound control signal from the mixed sound signal.

[0157] 17 is a block diagram showing the configuration of a sound signal processing unit 10F corresponding to the above-mentioned sound signal processing method. The sound signal processing unit 10F includes a line input unit 21F, a gain adjustment unit 22F, a mixer 23F, an early reflection sound control signal generation unit 214, an impulse response acquisition unit 151A, and a delay adjustment unit 28.

[0158] The line input unit 21F receives a plurality of line input signals and outputs them to the gain adjustment unit 22F. The gain adjustment unit 22F performs volume control of the plurality of line input signals. At this time, the gain adjustment unit 22F sets an individual gain for each of the plurality of line input signals and performs volume control. For example, the gain adjustment unit 22F sets an individual gain based on the level balance of the plurality of line input signals. The gain adjustment unit 22F outputs the plurality of volume-controlled line input signals to the mixer 23F.

[0159] The mixer 23F mixes a plurality of volume-controlled line input signals and outputs the mixed signal. The mixer 23F outputs the mixed signal to the early reflection sound control signal generating unit 214.

[0160] The early reflection sound control signal generation unit 214 convolves the mixing signal with an impulse response for the early reflection sound to generate an early reflection sound control signal. The early reflection sound control signal generation unit 214 outputs the early reflection sound control signal to the delay adjustment unit 28. As described above, the delay adjustment unit 28 adjusts the delay time of the early reflection sound control signal and outputs it to the speaker 51A. If there are multiple speakers, it is sufficient to include a matrix mixer 26 like the above-mentioned sound signal processing unit 10. The matrix mixer 26 distributes the early reflection sound control signal to the multiple speakers and outputs it.

[0161] With this configuration and method, the sound signal processing unit 10F can generate an early reflection sound control signal for a mixed signal obtained by mixing a plurality of line input signals, and can realize a desired sound field having a richer sound image and a wider spatial spread than ever before.

[0162] (3-4) An embodiment of the present invention in relation to the third embodiment is a sound signal processing method for adjusting the balance between the level of an early reflection sound control signal and the level of a sound signal that is the source of the early reflection sound control signal.

[0163] 18 is a block diagram showing the configuration of a sound signal processing unit 10G corresponding to the above-mentioned sound signal processing method. The sound signal processing unit 10G includes a line input unit 21G, a gain adjustment unit 22G, a mixer 23G, an early reflection sound control signal generation unit 214, a level setting unit 216, a level setting unit 217, an impulse response acquisition unit 151A, a level balance adjustment unit 153, and a delay adjustment unit 28.

[0164] The line input section 21G, the gain adjustment section 22G, and the mixer 23G are similar to the above-mentioned line input section 21F, the gain adjustment section 22F, and the mixer 23F, respectively. The mixer 23G outputs a mixed signal to the level setting section 216 and the level setting section 217.

[0165] The level balance adjustment unit 153 sets a gain for the direct sound and a gain for the early reflection sound using the level balance between the direct sound and the early reflection sound. The level balance adjustment unit 153 outputs the gain for the direct sound to the level setting unit 216 and outputs the gain for the early reflection sound to the level setting unit 217.

[0166] The level setting unit 216 controls the volume of the mixed signal by using the gain for the direct sound. The level setting unit 216 outputs the mixed signal, the volume of which has been controlled by the gain for the direct sound, to the synthesis unit 218.

[0167] The level setting unit 217 controls the volume of the mixed signal by using the gain for the early reflection sound. The mixed signal that has been volume-controlled by the gain for the early reflection sound is output to the early reflection sound control signal generating unit 214.

[0168] The early reflection sound control signal generating unit 214 convolves the impulse response for the early reflection sound with the mixing signal volume-controlled by the gain for the early reflection sound to generate an early reflection sound control signal. The early reflection sound control signal generating unit 214 outputs the early reflection sound control signal to the synthesis unit 218.

[0169] The synthesis unit 218 synthesizes the direct sound signal and the early reflection sound control signal, and outputs the result to the delay adjustment unit 28. As described above, the delay adjustment unit 28 adjusts the delay time of the synthesized signal, and outputs the signal to the speaker 51A. When there are multiple speakers, a matrix mixer 26 may be provided instead of the synthesis unit 218, as in the above-described sound signal processing unit 10. The matrix mixer 26 distributes and outputs the synthesized signal of the direct sound signal and the early reflection sound control signal to the multiple speakers. The matrix mixer 26 sets an allocation of the direct sound signal and the early reflection sound control signal for each speaker, and distributes and outputs the direct sound signal and the early reflection sound control signal to the multiple speakers using this allocation.

[0170] With this configuration and method, the sound signal processor 10G can adjust the level balance between the direct sound signal and the early reflection sound control signal, thereby realizing a desired sound field with a good balance between the direct sound and the early reflection sound, a rich sound image, and a spacious sound field.

[0171] (3-5) An embodiment of the third embodiment of the present invention is a sound signal processing method for generating a reverberation sound signal from a volume-controlled sound signal.

[0172] 19 is a block diagram showing the configuration of a sound signal processing unit 10H corresponding to the above-mentioned sound signal processing method. The sound signal processing unit 10H includes a line input unit 21H, a gain adjustment unit 22H, an early reflection sound control signal generation unit 214, a reverberation sound control signal generation unit 219, an impulse response acquisition unit 151A, and a delay adjustment unit 28.

[0173] The line input section 21H and the gain adjustment section 22H are respectively similar to the line input section 21E and the gain adjustment section 22E. The gain adjustment section 22H outputs a volume-controlled line input signal to the early reflection sound control signal generation section 214 and the reverberation sound control signal generation section 219. The early reflection sound control signal generation section 214 is also similar in configuration to the above-mentioned.

[0174] The reverberation sound control signal generator 219 convolves the volume-controlled line input signal with an impulse response for reverberation sound to generate a reverberation sound control signal. The reverberation sound control signal generator 219 outputs the reverberation sound control signal to the delay adjustment unit 28. As described above, the delay adjustment unit 28 adjusts the delay time of the reverberation sound control signal and outputs it to the speaker 61A. If there are multiple speakers, it is sufficient to include a matrix mixer 26 like the above-mentioned sound signal processor 10. The matrix mixer 26 distributes the reverberation sound control signal to the multiple speakers and outputs it.

[0175] With this configuration and method, the sound signal processor 10E can appropriately generate a reverberation control signal as well as an early reflection sound control signal, and can reproduce a desired sound field having a richer sound image and a wider spatial extent.

[0176] (3-6) One embodiment of the third embodiment of the present invention is a sound signal processing method that collects an output sound including a sound signal and generates a reverberation sound signal using the collected sound signal. That is, the sound signal processing unit collects a sound output from a speaker, feeds it back, and generates a reverberation sound signal from the collected sound signal.

[0177] With this configuration and method, the sound signal processing section can generate a reverberation signal that corresponds to the room 62B during performance, and can realize a desired sound field having a richer sound image and a wider spatial spread.

[0178] (3-7) One embodiment of the third embodiment of the present invention is a sound signal processing method in which volume control for reverberation is performed on the reverberation signal immediately before or immediately after the generation of the reverberation signal.

[0179] With this configuration and method, the sound signal processing unit can appropriately adjust the level of the reverberation sound, thereby enabling the sound signal processing unit to appropriately adjust, for example, the level balance between the early reflection sound and the reverberation sound, and the level balance between the direct sound and the reverberation sound.

[0180] (3-8) An embodiment of the third embodiment of the present invention is a sound signal processing method in which volume control for early reflection sounds is performed on the early reflection sound control signal immediately before or immediately after generation of the early reflection sound control signal.

[0181] With this configuration and method, the sound signal processing unit can appropriately adjust the level of the early reflection sound, thereby enabling the sound signal processing unit to appropriately adjust, for example, the level balance between the early reflection sound and the reverberation sound, and the level balance between the direct sound and the early reflection sound.

[0182] (3-9) One embodiment of the third embodiment of the present invention is a sound signal processing method that outputs a sound signal and an early reflection sound control signal together.

[0183] With this configuration and method, the sound signal processing unit can output the direct sound and the early reflected sound through the same (single) output system.

[0184] The description of the present embodiment is illustrative in all respects and is not restrictive. The scope of the present invention is indicated by the claims, not by the above-described embodiments. Furthermore, the scope of the present invention is intended to include all modifications within the meaning and scope of the claims. [Explanation of symbols]

[0185] 1, 1A, 1B...Sound field support system 10, 10A, 10B, 10C, 10D, 10E, 10F, 10G, 10H... Sound signal processing section 11A, 11B, 11C...Directional microphone 12A, 12B, 12C...Omnidirectional microphone 13A, 13B, 13C, 13D...Directional microphone 14A, 14B, 14C, 14D...Directional microphone 21, 21A, 21B...Sound signal acquisition section 21D, 21E, 21F, 21G, 21H...Line input section 22, 22E, 22F, 22G, 22H…Gain adjustment section 23, 23F, 23G...Mixer 24A…FIR filter 24B…FIR filter 25A…Level setting section 25B…Level setting section 26...Matrix mixer 27...Output section 28…Delay adjustment section 31…Memory 51A, 51B, 51C, 51D...Speakers 52A, 52B, 52C, 52D...Speakers 53A, 53B, 53C, 53D...Speakers 60…Stage 61, 611B, 612B, 613B…sound source 61A, 61B, 61C, 61D, 61E, 61F...Speakers Room 62, 62B... 151, 151A…Impulse response acquisition section 152...Level balance adjustment section 153...Level balance adjustment section 204A... Processing section 210...sound signal acquisition unit 211, 212...Level setting section 213…Synthetic section 214... Early reflection sound control signal generating unit 219...Reverberation control signal generating unit 230…Mixer 510D...directional microphone 620…Space

Claims

1. A plurality of sound signals are input through a line, generating a direct sound signal from the plurality of sound signals; The volume of each of the plurality of sound signals input through the line is controlled by an individual gain; mixing the plurality of volume-controlled sound signals and distributing the mixed sound signals to a plurality of signal processing systems for early reflection sounds and one signal processing system for reverberation sounds; generating an early reflection sound control signal for reproducing early reflection sounds in a predetermined space from sound signals of the plurality of signal processing systems for the early reflection sounds; Acquires sound signals from multiple omnidirectional microphones, synthesizing the sound signals acquired from the plurality of omnidirectional microphones and the sound signal of the one signal processing system for reverberation sound, and generating a reverberation sound control signal for reproducing the reverberation sound in the predetermined space using the synthesized signal; outputting the direct sound signal, the early reflection sound control signal, and the reverberation sound control signal, respectively; A method for processing a sound signal, comprising: The sound signals acquired from the plurality of omnidirectional microphones are not input to the sound signals of the plurality of signal processing systems for the early reflection sounds. Sound signal processing method.

2. adjusting the balance between the level of the early reflection sound control signal and the level of a sound signal that is a source of the early reflection sound control signal; The method for processing a sound signal according to claim 1 .

3. Immediately before or immediately after generating the reverberation control signal, a volume control for reverberation is performed on the reverberation control signal. The sound signal processing method according to claim 1 or 2.

4. Immediately before or immediately after the generation of the early reflection sound control signal, volume control for the early reflection sound is performed on the early reflection sound control signal. The sound signal processing method according to any one of claims 1 to 3.

5. outputting the direct sound signal and the early reflection sound control signal together; 5. The sound signal processing method according to claim 1.

6. A line input unit for inputting a plurality of sound signals through a line input; a sound signal processing unit that generates a direct sound signal from the plurality of sound signals; a volume control section that controls the volume of each of the plurality of line-input sound signals with an individual gain; and a mixer that mixes the plurality of volume-controlled sound signals and distributes the mixed sound signal to a plurality of signal processing systems for early reflection sounds and one signal processing system for reverberation sounds; an early reflection sound control signal generating unit that generates an early reflection sound control signal for reproducing an early reflection sound in a predetermined space from sound signals of the plurality of signal processing systems for the early reflection sound; A plurality of omnidirectional microphones for acquiring sound signals; a synthesis unit that synthesizes sound signals acquired from the plurality of omnidirectional microphones and a sound signal of one signal processing system for reverberation; a reverberation control signal generating unit that generates a reverberation control signal for reproducing a reverberation sound in the predetermined space by using a synthesis signal synthesized by the synthesis unit; an output unit that outputs the direct sound signal, the early reflection sound control signal, and the reverberation sound control signal; A sound signal processing device comprising: The sound signals acquired from the plurality of omnidirectional microphones are not input to the sound signals of the plurality of signal processing systems for the early reflection sounds. Sound signal processing device.

7. a level balance adjustment unit that adjusts the balance between a level of the early reflection sound control signal and a level of a sound signal that is a source of the early reflection sound control signal; The sound signal processing device according to claim 6 .

8. a level setting unit for reverberation that performs a volume control for reverberation on the reverberation control signal immediately before or immediately after generation of the reverberation control signal; The sound signal processing device according to claim 6 or 7.

9. a level setting unit for early reflection sounds that performs volume control for the early reflection sound control signal immediately before or immediately after generation of the early reflection sound control signal; The sound signal processing device according to any one of claims 6 to 8.

10. a speaker that outputs the direct sound signal and the early reflection sound control signal together; The sound signal processing device according to any one of claims 6 to 9.

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