Electronic musical instruments

Abstract

To adjust an amplitude spectrum of sound reaching a listening point from multiple loudspeakers of an electronic musical instrument to a flat characteristic over a wide range within an audible bandwidth.SOLUTION: An electronic musical instrument 100 comprises: a keyboard for receiving a playing operation; a first speaker 31 installed above the keyboard; a second speaker 32 installed below the keyboard 11; a signal generator 41 for generating an original signal Q0 in response to a playing operation received by the keyboard; a signal processor 42 for generating a first sound signal Q1 and a second sound signal Q2 by signal processing with respect to the original signal Q0; and a driving unit 43 for driving the first speaker 31 to radiate a first sound in response to the first sound signal Q1, and driving the second speaker 32 to radiate a second sound in response to the second sound signal Q2. The signal processing includes adjustment processing for bringing a phase of the first sound at a predetermined listening point and a phase of the second sound at the listening point closer to each other within the audible bandwidth by adjusting the phase of at least one of the first sound signal Q1 and the second sound signal Q2.SELECTED DRAWING: Figure 3

Description

Technical Field

[0001] The present disclosure relates to electronic musical instruments.

Background Art

[0002] Various techniques for controlling the frequency characteristics of acoustic signals have been proposed conventionally. For example, Patent Document 1 discloses a technique for controlling the latency and frequency response (amplitude characteristics and phase characteristics) of a FIR (Finite Impulse Response) filter.

Prior Art Documents

Patent Documents

[0003]

Patent Document 1

Summary of the Invention

Problems to be Solved by the Invention

[0004] By the way, in an electronic musical instrument such as a keyboard instrument, for example, sounds corresponding to performance operations by a performer are radiated from a plurality of speakers. The transmission characteristics until the sound radiated from the speaker reaches the ears of a listener such as the performer differ for each speaker. Therefore, the sounds reaching the listener's ears from each speaker may interfere with each other, and there is a possibility that the sound pressure perceived by the listener locally decreases at a specific frequency. Considering the above circumstances, one aspect of the present disclosure aims to adjust the amplitude spectrum of the sound reaching the listening point from a plurality of speakers to have flat characteristics over a wide range within the audible band.

Means for Solving the Problems

[0005] To solve the above problems, an electronic musical instrument according to one aspect of the present disclosure comprises: an operation receiving unit for receiving performance operations; a first speaker installed above the operation receiving unit; a second speaker installed below the operation receiving unit; a signal generation unit for generating a raw signal corresponding to the performance operations received by the operation receiving unit; a signal processing unit for generating a first tone signal and a second tone signal by signal processing of the raw signal; and a drive unit for driving the first speaker to radiate a first tone corresponding to the first tone signal and driving the second speaker to radiate a second tone corresponding to the second tone signal, wherein the signal processing includes an adjustment process that brings the phase of the first tone and the phase of the second tone at a predetermined listening point closer to each other within the audible band by adjusting the phase of at least one of the first tone signal and the second tone signal.

[0006] An electronic musical instrument according to another aspect of the present disclosure comprises: an operation receiving unit for receiving performance operations; a signal generation unit for generating a raw signal corresponding to the performance operations received by the operation receiving unit; a signal processing unit for generating a first tone signal and a second tone signal by signal processing of the raw signal; and a drive unit for driving a first speaker installed above the operation receiving unit to radiate a first tone corresponding to the first tone signal, and a second speaker installed below the operation receiving unit to radiate a second tone corresponding to the second tone signal, wherein the signal processing includes an adjustment process that brings the phase of the first tone and the phase of the second tone at a predetermined listening point closer to each other within the audible band by adjusting the phase of at least one of the first tone signal and the second tone signal. [Brief explanation of the drawing]

[0007] [Figure 1] These are front and side views of an electronic musical instrument. [Figure 2] This is a block diagram illustrating the electrical configuration of an electronic musical instrument. [Figure 3] This is a block diagram illustrating the functional configuration of an electronic musical instrument. [Figure 4] This is an explanatory diagram of the first and second frequency bands. [Figure 5] This is the amplitude spectrum observed at the listening point. [Figure 6] This is a block diagram of an FIR filter. [Figure 7] This is a flowchart of the processes performed by the control unit. [Figure 8] This is an explanatory diagram of the effects of the embodiment. [Figure 9] This is a block diagram illustrating the configuration of the signal processing unit in the second embodiment. [Modes for carrying out the invention]

[0008] A: First Embodiment Figure 1 shows a front view and a side view of the electronic instrument 100 according to the first embodiment. The electronic instrument 100 is an electronic keyboard instrument comprising a keyboard 11 and a housing 12. In the following description, we assume three mutually orthogonal axes (X axis, Y axis, Z axis). The X axis is the axis extending in the left-right direction (width direction) of the electronic instrument 100. The Y axis is the axis extending in the front-back direction (depth direction) of the electronic instrument 100. The Z axis is the axis extending in the up-down direction (height direction) of the electronic instrument 100. The Z axis corresponds to the vertical axis.

[0009] The keyboard 11 consists of multiple keys 13 (white keys and black keys) corresponding to different pitches. The multiple keys 13 are arranged along the X-axis. That is, the direction of the X-axis is the longitudinal direction of the keyboard 11. The performer U plays a desired piece of music by sequentially operating each of the multiple keys 13. In other words, the keyboard 11 is an operation reception unit that receives performance operations from the performer U. Performance operations include, for example, pressing and releasing keys. In the following explanation, a reference plane C is assumed. The reference plane C is the plane of symmetry of the electronic instrument 100. The plane of symmetry is a hypothetical plane on which the electronic instrument 100 is symmetrical. Specifically, the reference plane C is a hypothetical plane perpendicular to the X-axis and passes through the midpoint of the keyboard 11 in the direction of the X-axis.

[0010] The enclosure 12 is a structure that supports the keyboard 11. Specifically, the enclosure 12 comprises a right armrest 121, a left armrest 122, a shelf 123, an upper front panel 124, a lower front panel 125, and a top panel 126 (roof). The shelf 123 is a plate-shaped member that supports the keyboard 11 from below. The keyboard 11 and the shelf 123 are installed between the right armrest 121 and the left armrest 122. The upper front panel 124 and the lower front panel 125 are flat plates that constitute the front of the enclosure 12 and are installed parallel to the XZ plane. The upper front panel 124 is located above the keyboard 11, and the lower front panel 125 is located below the keyboard 11. The top panel 126 is a flat plate that constitutes the top surface of the enclosure 12.

[0011] Figure 2 is a block diagram illustrating the electrical configuration of the electronic musical instrument 100. The electronic musical instrument 100 comprises a control device 21, a storage device 22, a detection device 23, and a playback device 24. The control device 21 and the storage device 22 constitute a control system 20 that controls the operation of the electronic musical instrument 100. In the first embodiment, the control system 20 is shown mounted on the electronic musical instrument 100, but the control system 20 may be configured separately from the electronic musical instrument 100. For example, the control system 20 may be realized by an information device such as a smartphone or tablet terminal.

[0012] The control device 21 is one or more processors that control the operation of the electronic musical instrument 100. Specifically, the control device 21 is composed of one or more types of processors, such as a CPU (Central Processing Unit), GPU (Graphics Processing Unit), SPU (Sound Processing Unit), DSP (Digital Signal Processor), FPGA (Field Programmable Gate Array), or ASIC (Application Specific Integrated Circuit).

[0013] The storage device 22 is one or more memories that store the program executed by the control device 21 and various data used by the control device 21. For example, known recording media such as semiconductor recording media and magnetic recording media, or a combination of multiple types of recording media, can be used as the storage device 22. Alternatively, for example, a portable recording media that can be attached to and detached from the electronic musical instrument 100, or a recording media that the control device 21 can access via a communication network (e.g., cloud storage), may be used as the storage device 22.

[0014] The detection device 23 is a sensor unit that detects operations performed by the performer U on the keyboard 11. Specifically, the detection device 23 outputs performance information E that specifies the key 13 operated by the performer U from among the multiple keys 13 that make up the keyboard 11. The performance information E is, for example, MIDI (Musical Instrument Digital Interface) event data that specifies the number corresponding to the key 13 operated by the performer U.

[0015] The playback device 24 emits sound corresponding to the player U's input to the keyboard 11. As illustrated in Figure 3, the playback device 24 comprises a first speaker 31 and a second speaker 32. The first speaker 31 and the second speaker 32 are installed in the housing 12. As illustrated in Figure 1, the first speaker 31 is installed on the upper front panel 124 of the housing 12. That is, the first speaker 31 is positioned higher than the keyboard 11 in the Z-axis direction. On the other hand, the second speaker 32 is installed on the lower front panel 125 of the housing 12. That is, the second speaker 32 is positioned lower than the keyboard 11 in the Z-axis direction. As can be understood from the above explanation, the first speaker 31 is installed above the keyboard 11, and the second speaker 32 is installed below the keyboard 11. In other words, the keyboard 11 is located between the first speaker 31 and the second speaker 32 when viewed from the direction of the Y axis. In the following explanation, unless there is a need to distinguish between the first speaker 31 and the second speaker 32, they will both be collectively referred to as "speaker 3".

[0016] The first speaker 31 is a stereo speaker including a first left speaker 31L and a first right speaker 31R. The first left speaker 31L and the first right speaker 31R are installed on the upper front panel 124 with the central axis of the diaphragm parallel to the Y-axis. The first left speaker 31L and the first right speaker 31R are installed at intervals in the X-axis direction. Specifically, when viewed from the front of the electronic musical instrument 100, the first left speaker 31L is located on the left side of the reference plane C, and the first right speaker 31R is located on the right side of the reference plane C. The first left speaker 31L and the first right speaker 31R are located at an equal distance from the reference plane C.

[0017] The second speaker 32 is a stereo speaker including a second left speaker 32L and a second right speaker 32R. The second left speaker 32L and the second right speaker 32R are installed on the lower front panel 125 with the central axis of the diaphragm parallel to the Y-axis. The second left speaker 32L and the second right speaker 32R are installed at intervals in the X-axis direction on the lower front panel 125. Specifically, when viewed from the front of the electronic musical instrument 100, the second left speaker 32L is located on the left side of the reference plane C, and the second right speaker 32R is located on the right side of the reference plane C. The second left speaker 32L and the second right speaker 32R are located at an equal distance from the reference plane C.

[0018] The space (upper front panel 124) where the first left speaker 31L and the first right speaker 31R are installed is narrower than the space (lower front panel 125) where the second left speaker 32L and the second right speaker 32R are installed. Considering the above circumstances, the first left speaker 31L and the first right speaker 31R have a smaller diameter compared to the second left speaker 32L and the second right speaker 32R. As illustrated in FIG. 4, the frequency band (hereinafter referred to as "first frequency band") B1 of the sound that the first speaker 31 can radiate and the frequency band (hereinafter referred to as "second frequency band") B2 of the sound that the second speaker 32 can radiate are different. The first frequency band B1 and the second frequency band B2 are bands within the audible range. The audible range is, for example, in the range of 20 Hz or more and 20 kHz or less.

[0019] The first frequency band B1 is a band on the higher frequency side than the second frequency band B2. The first frequency band B1 and the second frequency band B2 partially overlap each other. That is, a part of the first frequency band B1 and a part of the second frequency band B2 overlap each other. Specifically, the frequency b1 at the lower end of the first frequency band B1 is located within the second frequency band B2, and the frequency b2 at the upper end of the second frequency band B2 is located within the first frequency band B1. In FIG. 4, a frequency band (hereinafter referred to as "overlap band") W in which the first frequency band B1 and the second frequency band B2 overlap each other is illustrated. The overlap band W is a frequency band with a predetermined width within the audible band.

[0020] As illustrated in FIG. 1, a point P (hereinafter referred to as "listening point") in a predetermined relationship with respect to the first speaker 31 and the second speaker 32 is assumed. The listening point P is a point where the sound radiated from each speaker 3 is mainly listened to, and is set at a position separated from the housing 12 of the electronic musical instrument 100. Specifically, the listening point P is a point corresponding to the head of the performer U who plays the electronic musical instrument 100. For example, the midpoint of both ear holes of a performer U with a standard height is illustrated as the listening point P. That is, the listening point P is located in front of the electronic musical instrument 100 (in the positive direction of the Y axis).

[0021] The position of the listening point P on the X-axis is within the range Rx. Range Rx is a range with a width of approximately 1m including the reference plane C (for example, reference plane C ± 0.5m). For example, the listening point P is located on the reference plane C. The position of the listening point P on the Y-axis is within the range Ry, which is spaced away from the electronic instrument 100 in the positive direction of the Y-axis. Range Ry is, for example, a range where the distance from the front end of the keyboard 11 is 1m or less. The front end of the keyboard 11 is the end of each white key located in the positive direction of the Y-axis. The position of the listening point P on the Z-axis is within the range Rz, which is spaced away from the floor surface on which the electronic instrument 100 is installed (hereinafter referred to as the "installation surface") in the positive direction of the Z-axis. Range Rz is, for example, a range where the distance from the installation surface is 1m or more and 1.5m or less. In the measurement of sound for adjusting the phase, etc., which will be described later, the average of the observed values ​​at multiple listening points placed within the above ranges (Rx, Ry, Rz) may be taken. In this case, the observed values ​​may be weighted and averaged so that the measurements taken at listening points closer to the center of the range are given more weight. Alternatively, multiple listening points may be arranged so that the center of the range is densely packed.

[0022] As explained above, the listening point P is located above the keyboard 11. That is, the listening point P is higher than the surface of the keyboard 11. Therefore, the distance D1 from the first speaker 31 to the listening point P is different from the distance D2 from the second speaker 32 to the listening point P. Specifically, distance D2 is greater than distance D1 (D2 > D1).

[0023] As illustrated in Figure 3, the control device 21 executes a program stored in the memory device 22 to realize multiple functions (signal generation unit 41, signal processing unit 42, drive unit 43) for emitting sound in response to the performance operations of the performer U.

[0024] The signal generation unit 41 generates the original signal Q0. The original signal Q0 is an audio signal corresponding to the performance operation received by the keyboard 11 from the performer U. The original signal Q0 is a stereo signal consisting of two channels, left and right. For example, the signal generation unit 41 is a MIDI sound source that generates the original signal Q0 according to the performance information E output by the detection device 23. That is, the original signal Q0 is a signal that represents the waveform of a sound with a pitch of 1 or more corresponding to the key 13 operated by the performer U. The signal generation unit 41 is a software sound source realized by, for example, the control device 21 executing a sound source program, or a hardware sound source realized by an electronic circuit dedicated to generating the original signal Q0.

[0025] The signal processing unit 42 generates a first tone signal Q1 and a second tone signal Q2 by signal processing on the original signal Q0. The first tone signal Q1 is a stereo signal consisting of two channels, left and right. Similarly, the second tone signal Q2 is a stereo signal consisting of two channels, left and right.

[0026] The drive unit 43 drives the playback device 24. Specifically, the drive unit 43 drives the first speaker 31 to radiate a sound corresponding to the first tone signal Q1 (hereinafter referred to as the "first tone"). The drive unit 43 also drives the second speaker 32 to radiate a sound corresponding to the second tone signal Q2 (hereinafter referred to as the "second tone").

[0027] Specifically, the drive unit 43 comprises a D / A converter that converts the first tone signal Q1 and the second tone signal Q2 from digital to analog, and an amplifier that amplifies the first tone signal Q1 and the second tone signal Q2. The drive unit 43 outputs a first output signal O1 generated from the first tone signal Q1 to the first speaker 31, and outputs a second output signal O2 generated from the second tone signal Q2 to the second speaker 32. The first output signal O1 is a two-channel stereo signal representing the waveform of the first tone, and the second output signal O2 is a two-channel stereo signal representing the waveform of the second tone. As illustrated above, the first speaker 31 radiates the first tone in the first frequency band B1, and the second speaker 32 radiates the second tone in the second frequency band B2.

[0028] Figure 5 shows the amplitude spectrum of the sound observed at the listening point P. Figure 5 illustrates the amplitude spectrum F1 of the first sound reaching the listening point P from the first speaker 31, and the amplitude spectrum F2 of the second sound reaching the listening point P from the second speaker 32. Proportionality 1 is a configuration in which signal processing by the signal processing unit 42 is not performed. That is, in proportionality 1, the original signal Q0 is supplied in common to both the first speaker 31 and the second speaker 32.

[0029] The first sound emitted from the first speaker 31 is reflected off the surface of the upper front panel 124 or the surface of the keyboard 11, etc., before reaching the listening point P. On the other hand, the second sound emitted from the second speaker 32 is reflected off the surface of the lower front panel 125, the underside of the shelf 123 or the mounting surface, etc., before reaching the listening point P. In other words, due to the difference in the positional relationship (i.e., asymmetry) of each speaker 3 with respect to the keyboard 11, the transmission characteristics from the first speaker 31 to the listening point P and the transmission characteristics from the second speaker 32 to the listening point P are different. Also, as mentioned above, the distance D1 from the first speaker 31 to the listening point P and the distance D2 from the second speaker 32 to the listening point P are different. Furthermore, due to differences in diaphragm size or mechanical / electrical characteristics, the phase characteristics differ between the first speaker 31 and the second speaker 32. In other words, even when a common signal is supplied to both the first speaker 31 and the second speaker 32, the frequency characteristics of the sound emitted by the first speaker 31 and the sound emitted by the second speaker 32 will differ.

[0030] As described above, the phase characteristics of the first and second tones reaching the listening point P differ due to a combination of various factors, including the transmission characteristics from each speaker 3 to the listening point P, the distance from each speaker 3 to the listening point P, and the phase characteristics of the speaker 3 itself. In a proportional relationship 1 where a common original signal Q0 is supplied to the first speaker 31 and the second speaker 32, the phase characteristics of the first tone and the phase characteristics of the second tone at the listening point P differ significantly due to the above differences. The signal processing performed by the signal processing unit 42 of the first embodiment is a process that brings the phase of the first tone and the phase of the second tone at the listening point P closer to each other within the audible range by adjusting the phase of at least one of the first tone signal Q1 and the second tone signal Q2. The specific configuration and operation of the signal processing unit 42 are described in detail below.

[0031] The signal processing unit 42 in Figure 3 comprises a first filter 421 and a second filter 422. The first filter 421 generates a first tone signal Q1 by processing the original signal Q0. The second filter 422 generates a second tone signal Q2 by processing the original signal Q0. Each of the first filter 421 and the second filter 422 includes an FIR (Finite Impulse Response) filter 50 as illustrated in Figure 6.

[0032] The FIR filter 50 includes (N-1) delay units 51[1] to 51[N-1], N multiplier units 52[1] to 52[N], and one adder unit 53. Each delay unit 51[n] (n=1 to N-1) generates a signal A[n+1] by delaying a signal A[n] for a predetermined time. The original signal Q0 is input as signal A[1] to the first stage delay unit 51[1]. The signal A[n+1] output by the delay unit 51[n] is supplied to the next stage delay unit 51[n+1] and multiplier unit 52[n+1].

[0033] Each multiplier 52[n] (n=1~N) generates a signal B[n] by multiplying a signal A[n] by a coefficient K[n]. The adder 53 adds the N signals B[1]~B[N] generated by each multiplier 52[n]. The adder 53 of the first filter 421 generates the first tone signal Q1, and the adder 53 of the second filter 422 generates the second tone signal Q2. As can be understood from the above explanation, the FIR filter 50 is a filter that convolves multiple coefficients (i.e., predetermined frequency responses) onto the signal to be processed.

[0034] The signal processing performed by the signal processing unit 42 includes bandpass filtering, delay processing, and adjustment processing. Each of the bandpass filtering, delay processing, and adjustment processing is implemented by either or both of the first filter 421 and the second filter 422. In other words, the bandpass filtering, delay processing, and adjustment processing are implemented by the FIR filter 50 illustrated in Figure 6.

[0035] Bandwidth filtering is a process that differentiates the frequency band of the first tone signal Q1 from the frequency band of the second tone signal Q2. Specifically, bandwidth filtering includes the process of extracting a component of the first frequency band B1 from the first tone signal Q1 using a first filter 421, and the process of extracting a component of the second frequency band B2 from the second tone signal Q2 using a second filter 422. In addition to the above, bandwidth filtering also includes a process (equalization) to suppress peaks in the frequency response of the first speaker 31 and the second speaker 32. As mentioned above, the first frequency band B1 and the second frequency band B2 overlap with each other within the overlapping band W. Note that the frequency band of the first tone signal Q1 and the first frequency band B1 of the first speaker 31 may be different. Also, the frequency band of the second tone signal Q2 and the second frequency band B2 of the second speaker 32 may be different. Furthermore, the process of suppressing peaks in the frequency response of the first speaker 31 and the second speaker 32 may be performed as a separate signal processing from the bandpass filtering process.

[0036] Delay processing is a process that delays one of the first tone signal Q1 and the second tone signal Q2 relative to the other. Specifically, delay processing is a process to reduce the phase difference between the first tone and the second tone caused by the difference between the distance D1 from the first speaker 31 to the listening point P and the distance D2 from the second speaker 32 to the listening point P (hereinafter referred to as "distance difference"). Specifically, the phase difference between the first tone and the second tone caused by the distance difference is the slope of the phase difference in the overlapping bandwidth W. As mentioned above, the distance D2 from the second speaker 32 to the listening point P is greater than the distance D1 from the first speaker 31 to the listening point P (D2 > D1). Taking the above distance difference into consideration, in the delay processing of the first embodiment, the first tone signal Q1 is delayed relative to the second tone signal Q2 by a time length corresponding to the distance difference (D2 - D1).

[0037] The adjustment process involves adjusting the phase of at least one of the first tone signal Q1 and the second tone signal Q2 to bring the phase of the first tone and the phase of the second tone at listening point P closer to each other within the audible range. In other words, the small phase difference remaining in the two phases, whose slopes have been made almost the same by the delay process, approaches zero in the impulse response through the adjustment process. Specifically, the adjustment process involves adjusting the phase of at least one of the first tone signal Q1 and the second tone signal Q2 to bring the phase of the first tone and the phase of the second tone at listening point P closer to each other within the overlapping band W where the first frequency band B1 and the second frequency band B2 overlap. In other words, the adjustment process is a process of bringing the phase of the first tone and the phase of the second tone closer at each frequency within the overlapping band W.

[0038] The conditions for the bandpass filtering, delay processing, and adjustment processing are controlled according to N coefficients K[1] to K[N] in the first filter 421 and the second filter 422, respectively. In the first embodiment, the N coefficients K[1] to K[N] in the first filter 421 and the second filter 422 are set so that the following conditions 1 to 3 are satisfied. Specifically, at the design stage of the electronic musical instrument 100, the sound reaching the listening point P from the first speaker 31 and the second speaker 32 is observed, and the N coefficients K[1] to K[N] are set according to the observation results so that the following conditions 1 to 3 are satisfied. For example, the frequency response of the transmission system from the input of the original signal Q0 to the signal processing unit 42 to the listening point P (observation point) is observed. The N coefficients K[1] to K[N] are stored in the memory device 22. Condition 1 (bandwidth filtering): The frequency band of the first tone signal Q1 and the frequency band of the second tone signal Q2 are different. Condition 2 (Delay Processing): The first tone signal Q1 is delayed relative to the second tone signal Q2 by a delay time corresponding to the distance difference (D2-D1). Condition 3 (Adjustment Process): The phase of the first tone at listening point P and the phase of the second tone at listening point P approach each other (ideally coincide) within the overlapping frequency band W.

[0039] The specific procedure for setting N coefficients K[1] to K[N] is as follows: (1) Determine the characteristics of the bandpass filtering process and measure the frequency response of the two transfer systems. (2) For the phase components of the two observed frequency responses, the phase difference in the overlapping bandwidth W is calculated. (3) The side to which the delay is introduced and its duration are determined according to the slope of the phase difference. (4) Based on the overall flat phase difference after delay adjustment, an impulse response is determined that brings that difference closer to zero. (5) From the characteristics, delay characteristics, and impulse response of the bandpass filter, N coefficients K[1] to K[N] are determined.

[0040] Figure 7 is a flowchart of the process executed by the control device 21. For example, the process shown in Figure 7 is initiated when the performer U operates the keyboard 11.

[0041] When processing begins, the control device 21 (signal generation unit 41) generates a source signal Q0 corresponding to the performance operation by the performer U (S1). The control device 21 (signal processing unit 42) generates a first tone signal Q1 and a second tone signal Q2 by signal processing on the source signal Q0 (S2). The signal processing includes bandpass filtering, delay processing, and adjustment processing, as described above. The control device 21 (drive unit 43) drives the first speaker 31 to radiate a first tone corresponding to the first tone signal Q1, and drives the second speaker 32 to radiate a second tone corresponding to the second tone signal Q2 (S3).

[0042] The effects of the first embodiment will be explained with reference to Figure 5. As mentioned above, in proportionality 1, the phase characteristics of the first sound at listening point P and the phase characteristics of the second sound at listening point P are significantly different. Therefore, a specific frequency component of the first sound and the corresponding frequency component of the second sound mutually cancel each other out at listening point P, and the sound pressure perceived by the performer U may locally decrease at that frequency. In this situation, a large dip occurs in the volume heard by the performer U at a specific frequency, due to interference of sound from the upper and lower speakers.

[0043] In contrast to proportionality 1, in the first embodiment, an adjustment process is performed to bring the phase of the first sound reaching the listening point P from the first speaker 31 and the phase of the second sound reaching the listening point P from the second speaker 32 closer to each other within the audible range. Therefore, as can be seen from the amplitude spectrum Ft of the first embodiment, localized drops in sound pressure caused by interference between the first and second sounds are suppressed. That is, the amplitude spectrum Ft of the sounds reaching the listening point P from the first speaker 31 and the second speaker 32 can be adjusted to have flat characteristics over a wide range within the audible range. Therefore, dips in the volume heard by the performer U at specific frequencies can be reduced.

[0044] The first embodiment is a configuration in which signal processing, including adjustment processing, is realized by an FIR filter 50. On the other hand, proportionality 2 in Figure 5 is a configuration in which signal processing (specifically, bandpass filtering and delay processing) that does not include adjustment processing is realized by an IIR (Infinite Impulse Response) filter. In proportionality 2, the localized drop in sound pressure caused by interference between the first and second tones is improved compared to proportionality 1 by signal processing using an IIR filter. However, the phase cannot be freely adjusted with an IIR filter. Therefore, in proportionality 2, as shown by the arrow in Figure 5, a localized drop in sound pressure caused by interference between the first and second tones is still observed at the listening point P. In other words, proportionality 2 cannot sufficiently reduce the dip in the volume at a specific frequency heard by the performer U.

[0045] In the first embodiment, as described above, signal processing including adjustment processing is realized by the FIR filter 50. In the FIR filter 50, the phase for each frequency can be changed with high precision by adjusting N coefficients K[1] to K[N]. That is, it is possible to bring the phase of the first sound and the phase of the second sound closer with high precision over a wide range within the audible band. Therefore, as can be seen from Figure 5, according to the first embodiment, the localized drop in sound pressure caused by interference between the first sound and the second sound is suppressed compared to proportionality 2. In other words, according to the first embodiment, dips at specific frequencies in the volume heard by the performer U can be sufficiently reduced.

[0046] Furthermore, in the first embodiment, a delay is added to the first sound signal Q1 or the second sound signal Q2 for a time duration corresponding to the distance difference between the distance D1 from the first speaker 31 to the listening point P and the distance D2 from the second speaker 32 to the listening point P (delay processing). Therefore, despite the configuration in which distances D1 and D2 are different, it is possible to bring the phases of the first sound and the second sound closer with high precision at the listening point P.

[0047] Figure 8 shows the results of observing sound pressure in the space in front of the electronic instrument 100. Specifically, Figure 8 illustrates the distribution of sound pressure within the reference plane C for both proportionality 2 and the first embodiment. Observation results for multiple frequencies within the audible range (500Hz, 800Hz, 1200Hz) are also shown. The dark gradation region is the region where the absolute value of sound pressure is large within the negative range, and the light gradation region is the region where the absolute value of sound pressure is large within the positive range.

[0048] In the region α illustrated for proportionality 2, the change in sound pressure over time is sufficiently small compared to the surrounding region. That is, in proportionality 2, a region α occurs where the volume heard by the performer U is locally low. In contrast to proportionality 2, in the first embodiment, the sound pressure changes over time over a wide area around the listening point P. That is, the occurrence of a region α where the volume locally decreases is suppressed according to the first embodiment.

[0049] Furthermore, in proportionality 2, as can be seen particularly from the measurement results at 800 Hz, regions with high sound pressure (pale tones) and regions with low sound pressure (dense tones) are mixed on the circumference around the electronic instrument 100. In other words, in proportionality 2, the wavefront of the sound emitted by the electronic instrument 100 is in a disordered state. Therefore, there is a problem that the volume perceived by performer U changes even if the performer U's head moves only slightly. In contrast to proportionality 2, in the first embodiment, only either a region with high sound pressure or a region with low sound pressure exists on the circumference around the electronic instrument 100. In other words, in the first embodiment, the wavefront of the sound emitted by the electronic instrument 100 propagates in a concentrically aligned state. Therefore, the volume perceived by performer U does not change easily even if the performer U's head moves only slightly. Also, for example, a listener positioned behind performer U (in the positive direction of the Y axis) can hear the sound just like performer U.

[0050] B: Second Embodiment A second embodiment will now be described. For elements whose function is the same as in the first embodiment in each of the embodiments described below, the same reference numerals as in the first embodiment will be used, and detailed descriptions of each will be omitted as appropriate.

[0051] Figure 9 is a block diagram illustrating the configuration of the signal processing unit 42 in the second embodiment. The signal processing unit 42 in the second embodiment includes a delay unit 423 in addition to the first filter 421 and second filter 422 similar to those in the first embodiment. The delay unit 423 is an element that realizes the aforementioned delay processing. That is, the delay unit 423 reduces the phase difference between the first sound and the second sound caused by the distance difference between distance D1 and distance D2.

[0052] Specifically, the delay unit 423 delays the original signal Q0 by a delay time corresponding to the distance difference between distance D2 and distance D1 (D2-D1). For example, the larger the distance difference (D2-D1), the larger the delay time set by the delay unit 423. The original signal Q0 after the delay by the delay unit 423 is supplied to the first filter 421. In other words, the delay unit 423 delays the first tone signal Q1 relative to the second tone signal Q2.

[0053] As mentioned above, the difference in phase characteristics between the first and second tones at listening point P is influenced by a combination of various factors, including the difference in transmission characteristics from each speaker to listening point P, including the distance difference (D2-D1), and the difference in phase characteristics of each speaker itself. The FIR filter 50 in the second embodiment functions to reduce the phase difference between the first and second tones caused by the differences in transmission characteristics and the differences in phase characteristics of each speaker. In other words, in the first embodiment, both delay processing and adjustment processing are realized by the FIR filter 50, whereas in the second embodiment, delay processing is realized by the delay unit 423, and adjustment processing is realized by the FIR filter 50.

[0054] The same effects as in the first embodiment are achieved in the second embodiment. Furthermore, in the second embodiment, since the delay processing is performed by the delay unit 423, the size of the FIR filter 50 can be reduced. Specifically, the number of taps in the FIR filter 50 is reduced, and as a result, the consumption of resources (e.g., computational power, memory, and power) in the electronic instrument 100 is reduced.

[0055] C: Variant Specific modifications added to each of the embodiments exemplified above are shown below. Multiple embodiments arbitrarily selected from the embodiments described above and the modifications exemplified below may be merged as appropriate, to the extent that they do not contradict each other.

[0056] (1) In the above-described configurations, the electronic instrument 100 is shown as having a first speaker 31 and a second speaker 32, but configurations in which the electronic instrument 100 further has other speakers 3 are also conceivable. The number of speakers 3 (number of channels) that the electronic instrument 100 has is arbitrary. Of the multiple speakers 3, one speaker 3 located above the keyboard 11 corresponds to the "first speaker," and one speaker 3 located below the keyboard 11 corresponds to the "second speaker." The relationship between the phase of the sound reaching the listening point from the "first speaker" and the phase of the second sound reaching the listening point from the "second speaker" is not a concern, as long as the phase of the sound reaching the listening point from the other speakers 3 and the phase of the first or second sound are brought closer together by adjustment processing.

[0057] (2) In the above-described embodiments, an example was given in which the first speaker 31 has a smaller diameter than the second speaker 32, but the difference in diameter between the first speaker 31 and the second speaker 32 is arbitrary. For example, an embodiment in which the first speaker 31 has a larger diameter than the second speaker 32, or an embodiment in which the first speaker 31 and the second speaker 32 have the same diameter is also conceivable.

[0058] (3) The listening point P is not limited to the point corresponding to the performer U's head. For example, if we consider a case where a listener is positioned behind the performer U and is listening to the performance, the position corresponding to the listener's head is exemplified as the listening point P. As can be understood from the above examples, the listening point P is comprehensively expressed as the point where the first and second tones should be heard, and can also be expressed as a point in a predetermined positional relationship with the electronic instrument 100 (especially each speaker 3).

[0059] (4) In each of the embodiments described above, an example was given in which signal processing including bandpass filtering, delay processing, and adjustment processing is realized by the FIR filter 50. However, the above signal processing may be realized by other types of filters such as IIR filters. In other words, the aforementioned proportionality 2 is included in the scope of this disclosure.

[0060] (5) The electronic musical instruments to which this disclosure applies are not limited to the electronic keyboard instruments exemplified in the above-described embodiments. For example, this disclosure applies to various electronic musical instruments such as electronic percussion instruments or electronic wind instruments. The term "operation receiving section" is comprehensively expressed as an element that receives performance operations, and the keyboard 11 in the above-described embodiments is an example of an "operation receiving section". For example, in an electronic percussion instrument, the head that the performer U strikes with a striking member (e.g., a stick), or in an electronic wind instrument, the keys that the performer U operates, are included in the concept of an "operation receiving section".

[0061] (6) The functions of the electronic musical instrument 100 (control system 20) in each of the above-described forms are realized through the cooperation of one or more processors constituting the control device 21 and the program stored in the storage device 22, as described above. The programs exemplified above can be provided in a form stored on a computer-readable recording medium and installed on a computer. The recording medium is, for example, a non-transitory recording medium, such as an optical recording medium (optical disc) like a CD-ROM, but also includes any known form of recording medium such as a semiconductor recording medium or a magnetic recording medium. Note that a non-transitory recording medium includes any recording medium except for transient propagation signals (transitory, propagating signals), and volatile recording media are not excluded. Furthermore, in a configuration in which a distribution device distributes a program via a communication network, the recording medium that stores the program in the distribution device corresponds to the non-transitory recording medium described above.

[0062] D: Addendum From the forms exemplified above, the following configuration can be understood, for example.

[0063] An electronic musical instrument according to one aspect of the present disclosure (Aspect 1) comprises: an operation receiving unit for receiving performance operations; a first speaker installed above the operation receiving unit; a second speaker installed below the operation receiving unit; a signal generation unit for generating a raw signal corresponding to the performance operations received by the operation receiving unit; a signal processing unit for generating a first tone signal and a second tone signal by signal processing of the raw signal; and a drive unit for driving the first speaker to radiate a first tone corresponding to the first tone signal and driving the second speaker to radiate a second tone corresponding to the second tone signal, wherein the signal processing includes an adjustment process that brings the phase of the first tone and the phase of the second tone at a predetermined listening point closer to each other within the audible band by adjusting the phase of at least one of the first tone signal and the second tone signal.

[0064] In the above embodiment, an adjustment process is performed to bring the phase of the first sound reaching the listening point from the first speaker and the phase of the second sound reaching the listening point from the second speaker closer to each other within the audible range. Therefore, the possibility of a localized decrease in sound pressure at a specific frequency due to interference between the first and second sounds at the listening point is reduced. In other words, the amplitude spectra of the sounds reaching the listening point from the first and second speakers can be adjusted to have flat characteristics over a wide range within the audible range.

[0065] The "operation reception unit" includes, for example, the keys in an electronic keyboard instrument that receive input from the user, the heads in an electronic percussion instrument that the user strikes with a striking object (e.g., a stick), or the keys in an electronic wind instrument that the user operates.

[0066] "Above the control receiver" means that the position of the first speaker in the vertical direction is higher than the control receiver, and is not limited to being directly above the control receiver. Similarly, "below the control receiver" means that the position of the second speaker in the vertical direction is lower than the control receiver, and is not limited to being directly below the control receiver. Sound radiated from the first speaker is reflected, for example, by the control receiver or an element located above it, before reaching the listening point. Similarly, sound radiated from the second speaker is reflected, for example, by the control receiver or an element located below it, before reaching the listening point.

[0067] The "original signal" is a signal that represents the waveform of sound in response to an operation performed on the control receiver. For example, a signal that represents the waveform of sound at a pitch specified by an operation on the control receiver, or a signal that represents the waveform of sound generated as a result of an operation on the control receiver, are examples of "original signals." Operation signals that represent the content of the performance operation (e.g., MIDI signals) are also included in the concept of "original signals."

[0068] The first and second tonal signals are two separate signals generated from the original signal. The relationship between the first and second tonal signals is arbitrary. For example, the first and second tonal signals may have different frequency bands. However, the first and second tonal signals may also share a common frequency characteristic.

[0069] A "listening point" is a point in a specific spatial or planar position relative to an electronic musical instrument. Specifically, examples of "listening points" include the location of the performer's head when performing the instrument, and the location of the listener's head when listening to the sound of the electronic instrument (for example, a point behind the performer). For example, the location of the performer's or listener's ear canal, or the midpoint between the left and right ear canals, can be considered "listening points." For example, in the case of a keyboard instrument, the "listening point" is assumed to be in a plane that passes through the center of the longitudinal direction of the keyboard and is perpendicular to that longitudinal direction.

[0070] In a specific example of Embodiment 1 (Embodiment 2), the operation reception unit receives performance operations from the performer, and the listening point is a point corresponding to the performer's head. According to the above embodiment, the amplitude spectrum of the sound heard by the performer of the electronic instrument can be adjusted to have flat characteristics over a wide range within the audible band.

[0071] In a specific example of Embodiment 1 or Embodiment 2 (Embodiment 3), the adjustment process is implemented using an FIR filter. Compared to an IIR filter, for example, an FIR filter can change the phase of the signal at each frequency with high precision. Therefore, in the form in which the adjustment process is implemented using an FIR filter, it is possible to bring the phases of the first tone and the second tone closer together with high precision over a wide range within the audible band. An FIR filter is a filter that convolves multiple coefficients (i.e., predetermined frequency responses) onto the signal to be processed. Multiple coefficients of the FIR filter are set so that the phases of the first tone and the second tone approach each other at the listening point.

[0072] In any specific example of Embodiments 1 to 3 (Embodiment 4), the first distance from the first speaker to the listening point is different from the second distance from the second speaker to the listening point, and the signal processing includes a delay process that delays either the first sound signal or the second sound signal. With this configuration, a delay is applied to either the first sound signal or the second sound signal for a time duration corresponding to the difference between the first distance from the first speaker to the listening point and the second distance from the second speaker to the listening point. Therefore, even in a configuration where the first and second distances are different, it is possible to bring the phases of the first sound and the second sound closer with high precision at the listening point.

[0073] The delay time due to delay processing is influenced by a combination of various factors, including the relationship between the phase characteristics of the first speaker and the second speaker, and the difference between the first and second distances.

[0074] In any specific example (5) of embodiments 1 to 4, the operation receiving unit is a keyboard with a plurality of keys arranged therein, the first speaker radiates the first tone in a first frequency band, the second speaker radiates the second tone in a second frequency band, the first frequency band and the second frequency band partially overlap with each other, and the adjustment process is a process of bringing the phase of the first tone at the listening point and the phase of the second tone at the listening point closer to each other within the overlapping band where the first frequency band and the second frequency band overlap with each other by adjusting the phase of at least one of the first tone signal and the second tone signal. With the above configuration, the amplitude spectrum of the musical sound heard by a keyboard instrument player in parallel with their performance can be adjusted to have flat characteristics over a wide range within the audible band.

[0075] An electronic musical instrument according to one aspect of the present disclosure (Aspect 6) comprises: an operation receiving unit for receiving performance operations; a signal generation unit for generating a raw signal corresponding to the performance operations received by the operation receiving unit; a signal processing unit for generating a first tone signal and a second tone signal by signal processing of the raw signal; and a drive unit for driving a first speaker installed above the operation receiving unit to radiate a first tone corresponding to the first tone signal, and a second speaker installed below the operation receiving unit to radiate a second tone corresponding to the second tone signal, wherein the signal processing includes an adjustment process that brings the phase of the first tone and the phase of the second tone at a predetermined listening point closer to each other within the audible band by adjusting the phase of at least one of the first tone signal and the second tone signal. [Explanation of symbols]

[0076] 100...Electronic instrument, 11...Keyboard, 12...Casing, 121...Right armrest, 122...Left armrest, 123...Shelf, 124...Upper front panel, 125...Lower front panel, 126...Top panel, 13...Key, 20...Control system, 21...Control device, 22...Memory device, 23...Detection device, 24...Playback device, 31...First speaker, 31L...First left speaker, 31R...First right speaker Pika, 32...Second speaker, 32L...Second left speaker, 32R...Second right speaker, 41...Signal generation unit, 42...Signal processing unit, 421...First filter, 422...Second filter, 423...Delay unit, 43...Drive unit, 50...FIR filter, 51[1]~51[N-1]...Delay unit, 52[1]~52[N]...Multiplication unit, 53...Addition unit.

Claims

1. An operation reception unit that receives performance instructions, A first speaker is installed above the aforementioned operation reception unit, A second speaker is installed below the aforementioned operation reception unit, The signal generation unit generates a signal corresponding to the performance operation received by the operation reception unit, A signal processing unit that generates a first tone signal and a second tone signal by performing signal processing on the original signal, The system comprises a drive unit that drives the first speaker to emit a first sound corresponding to the first sound signal, and a drive unit that drives the second speaker to emit a second sound corresponding to the second sound signal, The signal processing includes an adjustment process that brings the phase of the first sound and the phase of the second sound at a listening point within the predetermined range closer to each other within the audible range by adjusting the phase of at least one of the first sound signal and the second sound signal according to the average of the observed sound values ​​arriving from each of the first and second speakers at a plurality of points within a predetermined range. Electronic musical instrument.

2. The first distance from the first speaker to the listening point and the second distance from the second speaker to the listening point are different. The signal processing includes a delay process that delays either the first sound signal or the second sound signal. The electronic musical instrument according to claim 1.

3. The first speaker emits the first sound in the first frequency band, The second speaker emits the second sound in the second frequency band, The first frequency band and the second frequency band partially overlap with each other. The adjustment process involves adjusting the phase of at least one of the first sound signal and the second sound signal to bring the phase of the first sound at the listening point and the phase of the second sound at the listening point closer to each other within the overlapping frequency band where the first frequency band and the second frequency band overlap. An electronic musical instrument according to claim 1 or claim 2.

4. An operation reception unit that receives performance commands, The signal generation unit generates a signal corresponding to the performance operation received by the operation reception unit, A signal processing unit that generates a first tone signal and a second tone signal by signal processing of the original signal, The system comprises a drive unit that drives a first speaker, positioned above the operation receiving unit, to emit a first sound corresponding to the first sound signal, and a drive unit that drives a second speaker, positioned below the operation receiving unit, to emit a second sound corresponding to the second sound signal. The signal processing includes an adjustment process that brings the phase of the first sound and the phase of the second sound at a listening point within the predetermined range closer to each other within the audible range by adjusting the phase of at least one of the first sound signal and the second sound signal according to the average of the observed sound values ​​arriving from each of the first and second speakers at a plurality of points within a predetermined range. Electronic musical instrument.

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