WO2015025858A1 - Dispositif de haut-parleur et procédé de traitement de signal audio - Google Patents

Dispositif de haut-parleur et procédé de traitement de signal audio Download PDF

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Publication number
WO2015025858A1
WO2015025858A1 PCT/JP2014/071686 JP2014071686W WO2015025858A1 WO 2015025858 A1 WO2015025858 A1 WO 2015025858A1 JP 2014071686 W JP2014071686 W JP 2014071686W WO 2015025858 A1 WO2015025858 A1 WO 2015025858A1
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WIPO (PCT)
Prior art keywords
sound
audio signal
unit
input
channel
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PCT/JP2014/071686
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English (en)
Japanese (ja)
Inventor
真樹 片山
進 澤米
啓一 今岡
Original Assignee
ヤマハ株式会社
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Publication date
Priority claimed from JP2013269162A external-priority patent/JP6405628B2/ja
Priority claimed from JP2013269163A external-priority patent/JP6287191B2/ja
Priority claimed from JP2013272352A external-priority patent/JP6287202B2/ja
Priority claimed from JP2013272528A external-priority patent/JP6287203B2/ja
Application filed by ヤマハ株式会社 filed Critical ヤマハ株式会社
Priority to CN201480002397.6A priority Critical patent/CN104641659B/zh
Priority to EP14838464.7A priority patent/EP3038385B1/fr
Priority to US14/428,227 priority patent/US9674609B2/en
Publication of WO2015025858A1 publication Critical patent/WO2015025858A1/fr
Priority to US15/472,591 priority patent/US10038963B2/en

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    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04S—STEREOPHONIC SYSTEMS 
    • H04S7/00—Indicating arrangements; Control arrangements, e.g. balance control
    • H04S7/30—Control circuits for electronic adaptation of the sound field
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R2203/00—Details of circuits for transducers, loudspeakers or microphones covered by H04R3/00 but not provided for in any of its subgroups
    • H04R2203/12—Beamforming aspects for stereophonic sound reproduction with loudspeaker arrays
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04S—STEREOPHONIC SYSTEMS 
    • H04S2400/00—Details of stereophonic systems covered by H04S but not provided for in its groups
    • H04S2400/01—Multi-channel, i.e. more than two input channels, sound reproduction with two speakers wherein the multi-channel information is substantially preserved
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04S—STEREOPHONIC SYSTEMS 
    • H04S2420/00—Techniques used stereophonic systems covered by H04S but not provided for in its groups
    • H04S2420/01—Enhancing the perception of the sound image or of the spatial distribution using head related transfer functions [HRTF's] or equivalents thereof, e.g. interaural time difference [ITD] or interaural level difference [ILD]

Definitions

  • the present invention relates to a speaker device that outputs a sound beam having directivity and a sound that allows a virtual sound source to be perceived.
  • an array speaker device that outputs a sound beam having directivity by delaying an audio signal and distributing it to a plurality of speaker units (see Patent Document 1).
  • the array speaker device of Patent Document 1 is to localize a sound source by reflecting an audio beam of each channel to a wall and reaching from the periphery of a listener.
  • the array speaker device of Patent Document 1 performs a process of localizing a virtual sound source by performing a filter process based on a head-related transfer function for a channel that cannot reach an audio beam due to the shape of a room or the like. .
  • the array speaker device shown in Patent Document 1 changes the frequency characteristics by convolving a head-related transfer function corresponding to the shape of the head of the listener with the audio signal.
  • the listener perceives the virtual sound source by listening to the sound whose frequency characteristics have changed (sound that makes the virtual sound source perceive). As a result, the audio signal is virtually localized.
  • an array speaker device that outputs a sound beam having directivity by delaying an audio signal and distributing it to a plurality of speaker units (see, for example, Patent Documents 2 and 3).
  • the array speaker device of Patent Document 2 localizes a phantom sound source by outputting the same signal at a predetermined ratio using a C channel sound beam and a sound beam reflected on a wall and reaching a listener.
  • a phantom sound source is a virtual sound source that is localized in the middle of these different directions when the sound of the same channel is reached from different directions on the left and right sides of the listener.
  • the array speaker device of Patent Document 3 uses the sound beam reflected once by the left and right walls of the listener and the sound beam reflected twice by the left and right and rear walls, and the localization direction and surround of the front channel.
  • the phantom sound source is localized between the channel localization direction.
  • Japanese Unexamined Patent Publication No. 2008-227803 Japanese Unexamined Patent Publication No. 2005-159518 Japanese Unexamined Patent Publication No. 2010-213031
  • the virtual sound source cannot provide a sense of distance compared to the sound beam. Further, in the localization using the virtual sound source, if the listening position deviates from the specified position, the localization feeling becomes weak, so that the region where the localization feeling can be obtained is narrow. Further, since the head-related transfer function is set based on the shape of the model head, there is an individual difference in the sense of localization.
  • each sound beam does not completely match the volume or frequency characteristics of the beam reflected on the wall surface for each channel. Therefore, with a phantom sound source using an audio beam, it is difficult to clearly localize the sound source in the intended direction.
  • the array speaker device shown in Patent Document 1 exclusively localizes an audio signal and exclusively outputs an audio beam and a sound that perceives a virtual sound source only in a channel where the audio beam cannot reach. In order to improve the sense of localization, it is conceivable to output both a sound beam and a sound that makes a virtual sound source perceive at the same time.
  • the sound field effect is the effect of superimposing early reflections and rear reverberation sounds generated in an acoustic space such as a concert hall on the sound of the content. It makes the listener experience the realism of being in a space.
  • the initial reflected sound is the sound that is output from the sound source and that is reflected several times on the walls of the concert hall, etc., and reaches the listener, and is delayed from the direct sound that directly reaches the listener from the sound source. Reach the listener. Since the initial reflection sound has a smaller number of reflections than the rear reverberation sound, the reflection pattern is different for each direction of arrival. Therefore, the initial reflected sound has different frequency characteristics for each direction of arrival.
  • the rear reverberant sound is a sound that reaches the listener after being reflected on the wall of the concert hall more frequently than the initial reflected sound, and reaches the listener later than the initial reflected sound. Since the rear reverberant sound has a larger number of reflections than the initial reflected sound, the reflection pattern is substantially uniform regardless of the arrival direction. Therefore, the frequency components of the rear reverberation sound are substantially the same regardless of the arrival direction.
  • a sound simulating an actual early reflection sound is simply referred to as an initial reflection sound
  • a sound simulating an actual rear reverberation sound is simply referred to as a rear reverberation sound.
  • the sound that perceives the virtual sound source has a poor localization because the frequency characteristic of the head-related transfer function added to generate the virtual sound source changes when the initial reflected sound with different frequency characteristics for each direction of arrival is superimposed. Become clear.
  • the sound beam having directivity is superimposed on the rear reverberation sound having substantially the same frequency component regardless of the arrival direction, the audio signals of the respective channels tend to be similar, so that the sound images are combined, The localization is unclear.
  • the sound beam shown in Patent Document 1 may not be able to generate a surround sound field as desired by the listener.
  • the sound beam is difficult to reach the listener in an environment where the distance to the wall is long or an environment composed of a wall that hardly reflects the sound beam. This makes it difficult for the listener to perceive the sound source.
  • the method using the virtual sound source may not give sufficient localization feeling compared to the method using the sound beam.
  • the sense of localization tends to be weakened.
  • the method using the virtual sound source is based on the shape of the listener's head, there is an individual difference in the sense of localization.
  • an object of the present invention is to provide a speaker device that can clearly localize a sound source using localization by a virtual sound source while utilizing the characteristics of a sound beam.
  • Another object of the present invention is to provide a speaker device that can clearly locate a sound source in an intended direction even when an audio beam is used.
  • Another object of the present invention is to provide a speaker device that is more effective for allowing a listener to perceive a sound source than a conventional method using only an audio beam and a method using only a virtual sound source.
  • the speaker device of the present invention delays and distributes an input unit to which a plurality of channels of audio signals are input, a plurality of speakers, and a plurality of channels of audio signals input to the input unit to the plurality of speakers.
  • a directivity control unit for outputting a plurality of sound beams to the plurality of speakers, and performing a filtering process based on a head-related transfer function on at least one of the audio signals of the plurality of channels input to the input unit.
  • a localization adding unit for inputting to the speaker.
  • the audio signal processing method of the present invention includes: An input step in which multi-channel audio signals are input; A directivity control step of outputting a plurality of sound beams to the plurality of speakers by delaying and distributing the plurality of channels of audio signals input in the input step to the plurality of speakers; A localization adding step of performing filtering processing based on a head-related transfer function on at least one of the plurality of channels of audio signals input in the input step and inputting the filtered signals to the plurality of speakers.
  • the speaker device and the audio signal processing method of the present invention provide a sense of localization in both the sound beam and the virtual sound source, the sound source is clearly localized using the localization by the virtual sound source while utilizing the characteristics of the sound beam. Can do.
  • the speaker device and the audio signal processing method of the present invention can clearly localize a sound source in an intended direction even when an audio beam is used.
  • the speaker device and the audio signal processing method of the present invention even when a sound field effect is applied, the initial reflected sound characteristics having different frequency characteristics for each direction of arrival are not added to the sound perceived by the virtual sound source.
  • the frequency characteristic of the head-related transfer function is maintained, and the sense of localization is not impaired.
  • a sense of localization is given to both the sound beam and the virtual sound source. Therefore, the sense of localization is stronger than the conventional method using only the sound beam and the method using only the virtual sound source. Become.
  • FIG. 1 It is the schematic which shows the structure of AV system. It is a block diagram which shows the structure of an array speaker apparatus.
  • (A), (B) is a block diagram which shows the structure of a filter process part. It is a block diagram which shows the structure of a beamization process part.
  • (A), (B), and (C) are diagrams showing the relationship between an audio beam and channel setting. It is a block diagram which shows the structure of a virtual processing part.
  • (A), (B) is a block diagram which shows the structure of a localization addition part and a correction
  • (A), (B), (C) is a figure explaining the sound field which an array speaker apparatus produces
  • (A) is a block diagram showing a configuration of an array speaker device according to Modification Example 1
  • (B) is a diagram showing a relationship between a master volume and a gain of the array speaker device according to Modification Example 1.
  • (A) is a block diagram which shows the structure of the array speaker apparatus which concerns on the modification 2
  • (B) is a figure which shows the relationship between time, a front level ratio, and a gain.
  • (A), (B) is a figure which shows the array speaker apparatus which concerns on the modification 3.
  • FIG. It is the schematic which shows the structure of AV system. It is a block diagram which shows the structure of an array speaker apparatus.
  • (A), (B) is a block diagram which shows the structure of a filter process part.
  • FIG. 1 It is a block diagram which shows the structure of a beamization process part.
  • A), (B), and (C) are diagrams showing the relationship between an audio beam and channel setting. It is a block diagram which shows the structure of a virtual processing part.
  • A), (B) is a block diagram which shows the structure of a localization addition part and a correction
  • (A), (B) is a figure explaining the sound field which an array speaker apparatus produces
  • (A), (B) is a figure explaining the sound field which the array speaker apparatus 1002 produces
  • FIG. It is a block diagram which shows the structure of a phantom process part. It is a figure explaining the sound field which an array speaker apparatus generates. It is a figure explaining the sound field which an array speaker apparatus generates.
  • (A), (B) is a figure which shows the array speaker apparatus which concerns on a modification. It is a figure for demonstrating the AV system provided with the array speaker apparatus. It is a part of block diagram of an array speaker apparatus and a subwoofer.
  • (A) and (B) are block diagrams of an initial reflected sound processing unit and a rear reflected sound processing unit. It is a schematic diagram which shows the example of the impulse response measured in the concert hall.
  • (A), (B) is a block diagram of a localization addition part and a correction
  • (A), (B) is a block diagram of a localization addition part and a correction
  • (A), (B) is a figure which shows the array speaker apparatus and speaker set which concern on the modification of an array speaker apparatus. It is a block diagram which shows the structure of the array speaker apparatus which concerns on a modification.
  • FIG. 1 is a schematic diagram of an AV system 1 including an array speaker device 2 according to the present embodiment.
  • the AV system 1 includes an array speaker device 2, a subwoofer 3, a television 4, and a microphone 7.
  • the array speaker device 2 is connected to the subwoofer 3 and the television 4.
  • the array speaker device 2 receives an audio signal corresponding to an image reproduced on the television 4 and an audio signal from a content player (not shown).
  • the array speaker device 2 includes, for example, a rectangular parallelepiped housing and is installed in the vicinity of the television 4 (lower portion of the display screen of the television 4).
  • the array speaker device 2 includes, for example, 16 speaker units 21A to 21P, a woofer 33L, and a woofer 33R on the front surface (the surface facing the listener).
  • the speaker units 21A to 21P, the woofer 33L, and the woofer 33R correspond to “a plurality of speakers” of the present invention.
  • Speaker units 21A to 21P are arranged in a line along the horizontal direction when viewed from the listener.
  • the speaker unit 21A is disposed on the leftmost side when viewed from the listener, and the speaker unit 21P is disposed on the rightmost side when viewed from the listener.
  • the woofer 33L is disposed further to the left of the speaker unit 21A.
  • the woofer 33R is disposed on the right side of the speaker unit 21P.
  • the number of speaker units is not limited to 16, and may be, for example, 8 or the like.
  • the arrangement is not limited to the example in which the arrangement is arranged in one row along the horizontal direction, and for example, the arrangement may be arranged in three rows along the horizontal direction.
  • the subwoofer 3 is installed in the vicinity of the array speaker device 2. In the example of FIG. 1, it is arranged on the left side of the array speaker device 2, but the installation position is not limited to this example.
  • the array speaker device 2 is connected with a microphone 7 for measuring listening environment.
  • the microphone 7 is installed at the listening position.
  • the microphone 7 is used when measuring the listening environment, and does not need to be installed when actually viewing the content.
  • FIG. 2 is a block diagram showing the configuration of the array speaker device 2.
  • the array speaker device 2 includes an input unit 11, a decoder 10, a filter processing unit 14, a filter processing unit 15, a beamization processing unit 20, an addition processing unit 32, an addition processing unit 70, a virtual processing unit 40, and a control unit 35. ing.
  • the input unit 11 includes an HDMI receiver 111, a DIR 112, and an A / D conversion unit 113.
  • the HDMI receiver 111 inputs an HDMI signal conforming to the HDMI standard and outputs it to the decoder 10.
  • the DIR 112 receives a digital audio signal (SPDIF) and outputs it to the decoder 10.
  • the A / D converter 113 receives an analog audio signal, converts it into a digital audio signal, and outputs it to the decoder 10.
  • the decoder 10 is a DSP and decodes the input signal.
  • the decoder 10 inputs signals in various formats such as AAC (registered trademark), Dolby Digital (registered trademark), DTS (registered trademark), MPEG-1 / 2, MPEG-2 multichannel, MP3, etc. It is converted into an audio signal (digital audio signal of FL channel, FR channel, C channel, SL channel, and SR channel. Hereinafter, when simply referred to as an audio signal, the digital audio signal is indicated) and output.
  • a thick solid line shown in FIG. 2 indicates a multi-channel audio signal.
  • the decoder 10 also has a function of expanding, for example, a stereo channel audio signal to a multi-channel audio signal.
  • the multi-channel audio signal output from the decoder 10 is input to the filter processing unit 14 and the filter processing unit 15.
  • the filter processing unit 14 extracts and outputs a band suitable for each speaker unit from the multi-channel audio signal output from the decoder 10.
  • FIG. 3A is a block diagram illustrating the configuration of the filter processing unit 14, and FIG. 3B is a block diagram illustrating the configuration of the filter processing unit 15.
  • the filter processing unit 14 includes an HPF 14FL, an HPF 14FR, an HPF 14C, an HPF 14SL, and an HPF 14SR that input digital audio signals of the FL channel, the FR channel, the C channel, the SL channel, and the SR channel, respectively. Further, the filter processing unit 14 includes LPF 15FL, LPF 15FR, LPF 15C, LPF 15SL, and LPF 15SR for inputting digital audio signals of FL channel, FR channel, C channel, SL channel, and SR channel, respectively.
  • HPF14FL, HPF14FR, HPF14C, HPF14SL, and HPF14SR extract and output the high frequency of the audio signal of each input channel.
  • the cut-off frequencies of HPF 14FL, HPF 14FR, HPF 14C, HPF 14SL, and HPF 14SR are set to match the lower limit (eg, 200 Hz) of the reproduction frequency of speaker units 21A to 21P.
  • the output signals of HPF 14 FL, HPF 14 FR, HPF 14 C, HPF 14 SL, and HPF 14 SR are output to beam forming processing unit 20.
  • LPF15FL, LPF15FR, LPF15C, LPF15SL, and LPF15SR each extract and output the low frequency (for example, less than 200 Hz) of the audio signal of each input channel.
  • the cut-off frequencies of the LPF 15FL, LPF 15FR, LPF 15C, LPF 15SL, and LPF 15SR correspond to the cut-off frequencies of the HPF 14FL, HPF 14FR, HPF 14C, HPF 14SL, and HPF 14SR (for example, 200 Hz).
  • LPF 15FL, LPF 15C, and LPF 15SL are added by the adder 16 to become an L channel audio signal.
  • the L channel audio signal is further input to the HPF 30L and the LPF 31L.
  • the HPF 30L extracts and outputs the high range of the input audio signal.
  • the LPF 31L extracts and outputs the low frequency range of the input audio signal.
  • the cutoff frequencies of the HPF 30L and the LPF 31L correspond to the crossover frequency (for example, 100 Hz) between the woofer 33L and the subwoofer 3.
  • the crossover frequency may be changed by the listener.
  • the output signals of LPF15FR, LPF15C, and LPF15SR are added by the adder 17 to become an R channel audio signal.
  • the R channel audio signal is further input to the HPF 30R and the LPF 31R.
  • the HPF 30R extracts and outputs the high range of the input audio signal.
  • the LPF 31R extracts and outputs the low frequency range of the input audio signal.
  • the cutoff frequency of the HPF 30R corresponds to the crossover frequency (for example, 100 Hz) between the woofer 33R and the subwoofer 3. As described above, the crossover frequency may be changed by the listener.
  • the audio signal output from the HPF 30L is input to the woofer 33L via the addition processing unit 32.
  • the audio signal output from the HPF 30R is input to the woofer 33R via the addition processing unit 32.
  • the audio signal output from the LPF 31L and the audio signal output from the LPF 31R are added to a monaural signal by the addition processing unit 70 and input to the subwoofer 3.
  • the addition processing unit 70 also receives the LFE channel, adds the audio signal output from the LPF 31L and the audio signal output from the LPF 31R, and outputs the result to the subwoofer 3.
  • the filter processing unit 15 includes HPF 40FL, HPF 40FR, HPF 40C, HPF 40SL, and HPF 40SR for inputting digital audio signals of FL channel, FR channel, C channel, SL channel, and SR channel, respectively.
  • the filter processing unit 15 includes LPF 41FL, LPF 41FR, LPF 41C, LPF 41SL, and LPF 41SR that input digital audio signals of FL channel, FR channel, C channel, SL channel, and SR channel, respectively.
  • HPF40FL, HPF40FR, HPF40C, HPF40SL, and HPF40SR each extract and output the high frequency of the input audio signal of each channel.
  • the cutoff frequency of HPF40FL, HPF40FR, HPF40C, HPF40SL, and HPF40SR corresponds to the crossover frequency (for example, 100 Hz) between the woofer 33R and the woofer 33L and the subwoofer 3. As described above, the crossover frequency may be changed by the listener. Further, the cutoff frequency of HPF40FL, HPF40FR, HPF40C, HPF40SL, and HPF40SR may be the same as the cutoff frequency of HPF14FL, HPF14FR, HPF14C, HPF14SL, and HPF14SR.
  • the filter processing unit 15 may be configured not to output the low frequency to the subwoofer 3 as an aspect including only the HPF 40FL, HPF 40FR, HPF 40C, HPF 40SL, and HPF 40SR. Audio signals output from the HPF 40FL, HPF 40FR, HPF 40C, HPF 40SL, and HPF 40SR are output to the virtual processing unit 40.
  • LPF41FL, LPF41FR, LPF41C, LPF41SL, and LPF41SR each extract and output the low frequency range of the input audio signal of each channel.
  • the cut-off frequencies of LPF 41FL, LPF 41FR, LPF 41C, LPF 41SL, and LPF 41SR correspond to the crossover frequency (for example, 100 Hz).
  • the audio signals output from the LPF 41FL, LPF 41FR, LPF 41C, LPF 41SL, and LPF 41SR are added to a monaural signal by the adder 171 and then input to the subwoofer 3 via the addition processing unit 70.
  • the addition processing unit 70 the audio signal output from the LPF 41FL, LPF 41FR, LPF 41C, LPF 41SL, and LPF 41SR, the audio signal output from the LPF 31R and LPF 31L, and the audio signal of the LFE channel described above are added.
  • the addition processing unit 70 may include a gain adjustment unit that changes the addition ratio of these signals.
  • FIG. 4 is a block diagram illustrating a configuration of the beam processing unit 20.
  • the beamization processing unit 20 includes a gain adjustment unit 18FL, a gain adjustment unit 18FR, a gain adjustment unit 18C, a gain adjustment unit 18SL, which input digital audio signals of FL channel, FR channel, C channel, SL channel, and SR channel, respectively. And a gain adjusting unit 18SR.
  • the gain adjusting unit 18FL, the gain adjusting unit 18FR, the gain adjusting unit 18C, the gain adjusting unit 18SL, and the gain adjusting unit 18SR adjust the volume level of the audio signal by adjusting the gain of the audio signal of each channel.
  • the gain-adjusted audio signals of the respective channels are input to the directivity control unit 91FL, directivity control unit 91FR, directivity control unit 91C, directivity control unit 91SL, and directivity control unit 91SR, respectively.
  • Directivity control unit 91FL, directivity control unit 91FR, directivity control unit 91C, directivity control unit 91SL, and directivity control unit 91SR distribute audio signals of each channel to speaker units 21A to 21P.
  • the distributed audio signals for the speaker units 21A to 21P are combined by the combining unit 92 and supplied to the speaker units 21A to 21P.
  • the directivity controller 91FL, the directivity controller 91FR, the directivity controller 91C, the directivity controller 91SL, and the directivity controller 91SR adjust the delay amount of the audio signal supplied to each speaker unit.
  • the sounds output from the speaker units 21A to 21P are strengthened with respect to each other at the same phase and output as a sound beam having directivity. For example, when sound is output from all the speakers at the same timing, a sound beam having directivity is output in front of the array speaker device 2.
  • the directivity control unit 91FL, the directivity control unit 91FR, the directivity control unit 91C, the directivity control unit 91SL, and the directivity control unit 91SR change the delay amount added to each audio signal, thereby outputting the sound beam. The direction can be changed.
  • the directivity control unit 91FL, the directivity control unit 91FR, the directivity control unit 91C, the directivity control unit 91SL, and the directivity control unit 91SR are such that the phases of the sounds output from the speaker units 21A to 21P are at predetermined positions. It is also possible to provide a delay amount so that the sound beam is focused and to form an audio beam that focuses at the predetermined position.
  • the sound beam can reach the listening position directly from the array speaker device 2 or reflected on an indoor wall or the like.
  • the sound beam of the C channel audio signal can be output in the front direction, and the sound beam of the C channel can reach from the front of the listening position.
  • the sound beams of the FL channel audio signal and the FR channel audio signal are output in the left-right direction of the array speaker device 2 and reflected on the walls existing on the left and right of the listening position, respectively, from the left direction and the right direction of the listening position. Can be reached.
  • the sound beams of the SL channel audio signal and the SR channel audio signal are output in the left-right direction and reflected twice on the wall existing on the left and right of the listening position and on the wall existing on the rear side, respectively. It can be reached from the right rear direction.
  • Such setting of the output direction of the sound beam can be automatically performed by measuring the listening environment using the microphone 7.
  • the control unit 35 displays the test signal ( For example, an audio beam composed of white noise) is output to the beam processing unit 20.
  • the control unit 35 emits an audio beam from the left direction (referred to as 0 degree direction) parallel to the front surface of the array speaker apparatus 2 to the right direction (referred to as 180 degree direction) parallel to the front face of the array speaker apparatus 2. Turn. When the sound beam is turned on the front surface of the array speaker device 2, the sound beam is reflected on the wall of the room R according to the turning angle ⁇ of the sound beam, and the sound beam is picked up by the microphone 7 at a predetermined angle. .
  • the control unit 35 analyzes the level of the audio signal input from the microphone 7 as follows.
  • the control unit 35 stores the level of the audio signal input from the microphone 7 in a memory (not shown) in association with the output angle of the audio beam. Then, the control unit 35 assigns each channel of the multi-channel audio signal and the output angle of the sound beam based on the peak level of the audio signal. For example, the control unit 35 detects a peak not less than a predetermined threshold in the collected sound data. The control unit 35 assigns the output angle of the sound beam at the highest level among these peaks as the output angle of the C channel sound beam. For example, in FIG. 5B, the angle ⁇ 3a at the highest level is assigned as the output angle of the sound beam of the C channel.
  • control unit 35 assigns the output angles of the sound beams of the SL channel and the SR channel for the peaks that are close to each other with the peak set for the C channel. For example, in FIG. 5B, an angle ⁇ 2a close to the C channel and close to the 0 ° direction is assigned as the SL channel sound beam, and an angle ⁇ 4a close to the C channel and close to the 180 ° direction is assigned to the SR channel sound beam. Assigned as the output angle. Further, the control unit 35 assigns the output angles of the sound beams of the FL channel and the FR channel for the outermost peak. For example, in the example of FIG.
  • the angle ⁇ 1a closest to the 0 degree direction is assigned as the FL channel sound beam
  • the angle ⁇ 5a closest to the 0 degree direction is assigned as the output angle of the FR channel sound beam.
  • the control unit 35 detects the level difference at which the sound beam of each channel reaches the listening position, and the beam for setting the output angle of the sound beam based on the peak of the level measured by the detection unit.
  • An angle setting means is realized.
  • the setting is made so that the sound beam reaches from the surroundings at the position of the listener (microphone 7).
  • FIG. 6 is a block diagram illustrating a configuration of the virtual processing unit 40.
  • the virtual processing unit 40 includes a level adjusting unit 43, a localization adding unit 42, a correcting unit 51, a delay processing unit 60L, and a delay processing unit 60R.
  • the level adjustment unit 43 includes a gain adjustment unit 43FL, a gain adjustment unit 43FR, a gain adjustment unit 43C, a gain adjustment unit 43SL, which input digital audio signals of the FL channel, the FR channel, the C channel, the SL channel, and the SR channel.
  • a gain adjustment unit 43SR is provided.
  • the gain adjusting unit 43FL, the gain adjusting unit 43FR, the gain adjusting unit 43C, the gain adjusting unit 43SL, and the gain adjusting unit 43SR adjust the level of the audio signal by adjusting the gain of the audio signal of each channel.
  • the gain of each gain adjustment unit is set based on the detection result of the test sound beam by the control unit 35 as setting means. For example, as shown in FIG. 5B, the C-channel sound beam is the highest level because it is a direct sound. Therefore, the gain of the gain adjusting unit 43C is set to the lowest. Further, the C-channel sound beam is a direct sound and is less likely to depend on the room environment, so may be a fixed value, for example.
  • a gain corresponding to the level difference from the C channel is set.
  • the gain of the gain adjustment unit 43FR is set to 0.4.
  • the SR channel detection level G2 0.4
  • the gain of the gain adjustment unit 43SR is set to 0.6. In this way, the gain of each channel is adjusted.
  • the control unit 35 rotates the sound beam of the test signal so that the sound beam of each channel reaches the listening position.
  • the listener instructs the control unit 35 to manually output a sound beam using a user interface (not shown), and the level difference at which the sound beam of each channel reaches the listening position. It is good also as an aspect which detects.
  • the gain adjustment unit 43FL, gain adjustment unit 43FR, gain adjustment unit 43C, gain adjustment unit 43SL, and gain adjustment unit 43SR are set for each channel separately from the level detected by sweeping the test audio beam. May be measured. Specifically, the test sound beam can be output for each channel in the direction determined by the sweep of the test sound beam, and the sound collected by the microphone 7 at the listening position can be analyzed.
  • the gain-adjusted audio signal of each channel is input to the localization adding unit 42.
  • the localization adding unit 42 performs a process of localizing the input audio signal of each channel as a virtual sound source at a predetermined position.
  • a head-related transfer function hereinafter referred to as HRTF
  • HRTF head-related transfer function
  • HRTF is an impulse response that expresses the loudness, arrival time, frequency characteristics, etc. from the virtual speaker installed at a certain position to the left and right ears.
  • the localization adding unit 42 can cause the listener to localize the virtual sound source by adding HRTF to the input audio signal of each channel and emitting sound from the woofer 33L or the woofer 33R.
  • FIG. 7A is a block diagram showing the configuration of the localization adding unit 42.
  • the localization adding unit 42 includes an FL filter 421L, an FR filter 422L, a C filter 423L, an SL filter 424L, and an SR filter 425L for convolving an HRTF impulse response with the audio signal of each channel, an FL filter 421R, an FR filter 422R, A C filter 423R, an SL filter 424R, and an SR filter 425R.
  • the audio signal of the FL channel is input to the FL filter 421L and the FL filter 421R.
  • the FL filter 421L adds an HRTF of the path from the position of the virtual sound source VSFL (see FIG. 8A) in front of the listener to the left ear to the FL channel audio signal.
  • the FL filter 421R adds an HRTF of a path from the position of the virtual sound source VSFL to the right ear to the audio signal of the FL channel.
  • an HRTF from the position of the virtual sound source around the listener to each ear is given.
  • the adder 426L synthesizes the audio signals to which the HRTF has been added by the FL filter 421L, the FR filter 422L, the C filter 423L, the SL filter 424L, and the SR filter 425L, and outputs the synthesized audio signal to the correcting unit 51.
  • the adder 426R synthesizes the audio signals to which the HRTF has been added by the FL filter 421R, the FR filter 422R, the C filter 423R, the SL filter 424R, and the SR filter 425R, and outputs the synthesized audio signal VR to the correcting unit 51.
  • the correction unit 51 performs a crosstalk cancellation process.
  • FIG. 7B is a block diagram illustrating a configuration of the correction unit 51.
  • the correction unit 51 includes a direct correction unit 511L, a direct correction unit 511R, a cross correction unit 512L, and a cross correction unit 512R.
  • the audio signal VL is input to the direct correction unit 511L and the cross correction unit 512L.
  • the audio signal VR is input to the direct correction unit 511R and the cross correction unit 512R.
  • the direct correction unit 511L performs processing to make the listener perceive that the sound output from the woofer 33L is emitted near the left ear.
  • a filter coefficient is set such that the frequency characteristic of the sound output from the woofer 33L is flat at the position of the left ear.
  • the direct correction unit 511L processes the input audio signal VL with the filter and outputs an audio signal VLD.
  • a filter coefficient is set such that the frequency characteristic of the sound output from the woofer 33R is flat at the position of the right ear.
  • the direct correction unit 511R processes the input audio signal VL with the filter and outputs the audio signal VRD.
  • the cross correction unit 512L is set with a filter coefficient for giving a frequency characteristic of a sound that circulates from the woofer 33L to the right ear.
  • the sound (VLC) that circulates from the woofer 33L to the right ear is reversed in phase by the synthesizer 52R and emitted from the woofer 33R, thereby suppressing the sound of the woofer 33L from being heard by the right ear. This makes the listener perceive that the sound emitted from the woofer 33R is emitted near the right ear.
  • the cross correction unit 512R is set with a filter coefficient for giving a frequency characteristic of a sound that circulates from the woofer 33R to the left ear.
  • the sound (VRC) that circulates from the woofer 33R to the left ear is reversed in phase by the synthesizer 52L and emitted from the woofer 33L, thereby suppressing the sound of the woofer 33R from being heard by the left ear. This makes the listener perceive that the sound emitted from the woofer 33L is emitted near the left ear.
  • the audio signal output from the synthesizing unit 52L is input to the delay processing unit 60L.
  • the audio signal delayed by a predetermined time by the delay processing unit 60L is input to the addition processing unit 32.
  • the audio signal output from the synthesis unit 52R is input to the delay processing unit 60R.
  • the audio signal delayed by a predetermined time by the delay processing unit 60R is input to the addition processing unit 32.
  • the delay time by the delay processing unit 60L and the delay processing unit 60R is set longer than the longest delay time among the delay times given by the directivity control unit of the beamization processing unit 20, for example. Thereby, the sound that perceives the virtual sound source does not hinder the formation of the sound beam.
  • a delay processing unit may be provided after the beam forming processing unit 20 so as to add a delay to the sound beam side so that the sound beam does not hinder the sound that localizes the virtual sound image.
  • the audio signal output from the delay processing unit 60L is input to the woofer 33L via the addition processing unit 32.
  • the addition processing unit 32 the audio signal output from the delay processing unit 60L and the audio signal output from the HPF 30L are added.
  • the addition processing unit 32 may include a configuration of a gain adjustment unit that changes the addition ratio of these audio signals.
  • the audio signal output from the delay processing unit 60R is input to the woofer 33R via the addition processing unit 32.
  • the audio signal output from the delay processing unit 60R and the audio signal output from the HPF 30R are added.
  • the addition processing unit 32 may include a gain adjustment unit that changes the addition ratio of these audio signals.
  • the solid line arrow indicates the path of the sound beam output from the array speaker device 2.
  • a white star indicates the position of the sound source generated by the sound beam
  • a black star indicates the position of the virtual sound source.
  • the array speaker device 2 outputs five sound beams in the same manner as the example shown in FIG.
  • the C-channel audio signal is set to an audio beam that is focused at a position behind the array speaker device 2.
  • the listener perceives that the sound source SC is in front of the listener.
  • an audio beam focused on the position of the left front wall of the room R is set, and the listener perceives that the sound source SFL is on the left front wall of the listener.
  • the audio signal of the FR channel is set with a sound beam that focuses on the position of the right front wall of the room R, and the listener perceives that the sound source SFR is on the right front wall of the listener.
  • the audio signal of the SL channel is set with an audio beam focused on the position of the left rear wall of the room R, and the listener perceives that the sound source SSL is on the left rear wall of the listener.
  • the audio signal of the SR channel is set with a sound beam that focuses on the position of the right rear wall of the room R, and the listener perceives that the sound source SSR is on the left rear wall of the listener.
  • the localization adding unit 42 sets the position of the virtual sound source at substantially the same position as the positions of the sound sources SFL, SFR, SC, SSL, and SSR. Therefore, the listener perceives the virtual sound sources VSC, VSFL, VSFR, VSSL, and VSSR at substantially the same positions as the positions of the sound sources SFL, SFR, SC, SSL, and SSR, as shown in FIG.
  • the position of the virtual sound source need not be set at the same position as the focal point of the sound beam, and may be a predetermined direction. For example, the virtual sound source VSFL is set to 30 degrees left, the virtual sound source VSFR is set to 30 degrees right, the virtual sound source VSSL is set to 120 degrees left, the virtual sound source VSSR is set to 120 degrees right, and the like.
  • the array speaker device 2 can supplement the sense of localization by the sound beam with the virtual sound source, and can improve the sense of localization as compared with the case of using only the sound beam or the case of using only the virtual sound source.
  • the sound source SSL and the sound source SSR of the SL channel and the SR channel are generated when the sound beam is reflected twice on the wall, and thus there is a case where a clear localization feeling cannot be obtained as compared with the channel on the front side.
  • the array speaker device 2 can supplement the sense of localization with the virtual sound source VSSL and the virtual sound source VSSR generated by the sound that directly reaches the ears of the listener by the woofer 33L and the woofer 33R, and the localization feeling of the SL channel and the SR channel can be compensated. There is no loss.
  • the control unit 35 of the array speaker device 2 detects the level difference at which the sound beam of each channel reaches the listening position, and the gain adjustment unit 43FL of the level adjustment unit 43 based on the detected level difference.
  • the levels of the gain adjustment unit 43FR, the gain adjustment unit 43C, the gain adjustment unit 43SL, and the gain adjustment unit 43SR are set. Thereby, the level (or level ratio) between each channel of the localization adding unit 42 and each channel of the sound beam is adjusted.
  • a curtain 501 with low acoustic reflectivity exists on the right wall of the room R in FIG. 8A, and the sound beam is difficult to reflect. Therefore, as shown in FIG. 8B, the peak level at the angle ⁇ a4 is lower than the peak levels at the other angles. In this case, the level of the sound beam that reaches the listening position of the SR channel is lower than that of the other channels.
  • the control unit 35 sets the gain of the gain adjustment unit 43SR higher than that of the other level adjustment units, and for the SR channel, sets the level of the localization addition unit higher than that of the other channels, so To strengthen the effect.
  • the control unit 35 sets the level ratio in the level adjustment unit 43 based on the level difference detected by the test sound beam. As a result, for the channel where the sense of localization due to the sound beam is low, the sense of localization is strongly supplemented by the virtual sound source. Even in this case, since the sound beam itself is output, the sense of localization by the sound beam remains, and only a specific channel becomes a virtual sound source, and there is no sense of incongruity, and the auditory connection between channels is maintained. Is done.
  • the array speaker device 2 estimates the angle at which the sound beam reaches even if the number of detected peaks is small relative to the number of channels, and It is preferable to assign an output angle of the sound beam. For example, in the example of FIG. 8C, no peak is detected at the angle to which the SR channel should be allocated, but the SR channel is set at an angle ⁇ a4 that is symmetrical with the angle ⁇ a2 with the angle ⁇ a3 being the highest level as the center angle. Allocating and outputting the SR channel sound beam. Then, the control unit 35 sets the gain of the gain adjustment unit 43SR high according to the level difference between the detection level G1 at the angle ⁇ a3 and the detection level G2 at the angle ⁇ a4.
  • the sound beam itself is output even for a channel for which the effect of adding localization by the virtual sound source is strongly set, so that the sound of the sound beam of the channel can be heard to some extent. Therefore, only a specific channel becomes a virtual sound source, and a sense of incongruity does not occur, and an auditory connection between channels is maintained.
  • the aspect in which the level ratio between each channel of the localization adding unit 42 and each channel of the sound beam is adjusted by adjusting the gain of each gain adjusting unit of the level adjusting unit 43 is described.
  • the gains of the gain adjusting unit 18FL, the gain adjusting unit 18FR, the gain adjusting unit 18C, the gain adjusting unit 18SL, and the gain adjusting unit 18SR in the beam processing unit 20 each channel of the localization adding unit and each of the sound beams are adjusted. It is good also as an aspect which adjusts a level ratio with a channel.
  • FIG. 9A is a block diagram showing a configuration of the array speaker device 2A according to the first modification. The same components as those in the array speaker device 2 shown in FIG.
  • the array speaker device 2A further includes a volume setting receiving unit 77.
  • the volume setting reception unit 77 receives a master volume setting from a listener.
  • the control unit 35 adjusts the gain of a power amplifier (not shown) (for example, an analog amplifier) according to the master volume setting received from the volume setting reception unit 77. Thereby, the volume of all the speaker units is changed collectively.
  • the control unit 35 sets the gains of all gain adjustment units in the level adjustment unit 43 in accordance with the master volume setting received from the volume setting reception unit 77. For example, as shown in FIG. 9B, the gains of all gain adjustment units in the level adjustment unit 43 are set higher as the master volume value becomes smaller. Thus, when the master volume setting is lowered, the level of the sound reflected from the wall of the sound beam is lowered, and the surround feeling may be lowered. Therefore, the control unit 35 maintains the surround feeling by setting the level of the localization adding unit 42 higher as the master volume value becomes smaller, and strengthening the localization addition effect by the virtual sound source.
  • FIG. 10A is a block diagram showing a configuration of the array speaker device 2B according to the second modification. The same components as those in the array speaker device 2 shown in FIG.
  • control unit 35 inputs the audio signal of each channel, and compares the level of the audio signal of each channel (functions as a comparison unit).
  • the control unit 35 dynamically sets the gain of each gain adjustment unit in the level adjustment unit 43 based on the comparison result.
  • the control unit 35 calculates the level ratio (front level ratio) between the front channel and the surround channel, and each gain in the level adjustment unit 43 according to the front level ratio. It is also possible to set the gain of the adjustment unit. That is, when the surround channel level is relatively high, the control unit 35 sets the gain of the level adjustment unit 43 (the gain adjustment unit 43SL and the gain adjustment unit 43SR) high, and the surround channel level is relatively low. In this case, the gain of the level adjustment unit 43 (gain adjustment unit 43SL and gain adjustment unit 43SR) is set low.
  • the level of the surround channel becomes relatively high, the effect of adding the localization by the virtual sound source is strengthened to emphasize the effect of the surround channel.
  • the level of the front channel is relatively high, the sound beam is set to a large level, and the effect of the front channel by the sound beam is emphasized, so that a sense of localization can be obtained compared to the virtual sound source localization. It is possible to make the listening area relatively wide.
  • the gain of the level adjusting unit 43 (the gain adjusting unit 43SL and the gain adjusting unit 43SR) is reduced when the level of the surround channel is relatively low, it may be more difficult to hear the surround channel by the sound beam.
  • the gain of the level adjustment unit 43 (gain adjustment unit 43SL and gain adjustment unit 43SR) is set high, and when the level of the surround channel is relatively high, the level adjustment unit 43 ( The gains of the gain adjusting unit 43SL and the gain adjusting unit 43SR may be reduced.
  • the level comparison between channels and the calculation of the level ratio between the front channel and the surround channel may be performed for the entire frequency band.
  • the audio signal of each channel is divided into predetermined bands and divided. It is also possible to compare the levels for each band or calculate the level ratio between the front channel and the surround channel. For example, since the lower limit of the reproduction frequency of each of the speaker units 21A to 21P for outputting the sound beam is 200 Hz, the level ratio between the front channel and the surround channel is calculated in a band of 200 Hz or higher.
  • FIG. 11A is a diagram showing an array speaker device 2C according to the third modification. A description of the same configuration as that of the array speaker device 2 is omitted.
  • Array speaker device 2C is different from array speaker device 2 in that sounds output from woofer 33L and woofer 33R are output from speaker unit 21A and speaker unit 21P, respectively.
  • the array speaker device 2C outputs sound that makes a virtual sound source perceived from the speaker units 21A and 21P at both ends of the speaker units 21A to 21P.
  • the speaker unit 21A and the speaker unit 21P are the speaker units arranged at the end of the array speaker, and are arranged on the left and right sides as viewed from the listener. Therefore, the speaker unit 21A and the speaker unit 21P are suitable for outputting the sound of the L channel and the R channel, respectively, and are suitable as the speaker unit for outputting the sound that makes the virtual sound source be perceived.
  • the array speaker device 2 does not have to include all the speaker units 21A to 21P, the woofer 33L, and the woofer 33R in a single housing.
  • each speaker unit may be provided in an individual casing, and the casings may be arranged side by side.
  • the input multi-channel audio signals are respectively delayed and distributed to a plurality of speakers, and the input multi-channel audio signals are filtered based on the head-related transfer function. Any aspect that inputs to a plurality of speakers belongs to the technical scope of the present invention.
  • FIG. 12 is a schematic diagram of an AV system 1001 including an array speaker device 1002 according to the second embodiment.
  • the AV system 1001 includes an array speaker device 1002, a subwoofer 1003, a television 1004, and a microphone 1007.
  • Array speaker apparatus 1002 is connected to subwoofer 1003 and television 1004.
  • the array speaker device 1002 receives an audio signal corresponding to an image reproduced on the television 1004 and an audio signal from a content player (not shown).
  • the array speaker device 1002 includes, for example, a rectangular parallelepiped housing, and is installed in the vicinity of the television 1004 (lower portion of the display screen of the television 1004).
  • the array speaker device 1002 includes, for example, 16 speaker units 1021A to 1021P, a woofer 1033L, and a woofer 1033R on the front surface (the surface facing the listener).
  • Speaker units 1021A to 1021P are arranged in a line along the horizontal direction as viewed from the listener.
  • the speaker unit 1021A is disposed on the leftmost side when viewed from the listener, and the speaker unit 1021P is disposed on the rightmost side when viewed from the listener.
  • the woofer 1033L is disposed further to the left of the speaker unit 1021A.
  • the woofer 1033R is disposed on the right side of the speaker unit 1021P.
  • the speaker units 1021A to 1021P, the woofer 1033L, and the woofer 1033R correspond to “a plurality of speakers” of the present invention.
  • the number of speaker units is not limited to 16, and may be, for example, 8 or the like.
  • the arrangement is not limited to the example in which the arrangement is arranged in one row along the horizontal direction, and for example, the arrangement may be arranged in three rows along the horizontal direction.
  • the subwoofer 1003 is installed in the vicinity of the array speaker device 1002.
  • the speaker is disposed on the left side of the array speaker device 1002, but the installation position is not limited to this example.
  • the array speaker device 1002 is connected to a microphone 1007 for measuring listening environment.
  • the microphone 1007 is installed at the listening position.
  • the microphone 1007 is used when measuring the listening environment, and does not need to be installed when actually viewing the content.
  • FIG. 13 is a block diagram showing the configuration of the array speaker apparatus 1002.
  • the array speaker apparatus 1002 includes an input unit 1011, a decoder 1010, a filter processing unit 1014, a filter processing unit 1015, a beam processing unit 1020, an addition processing unit 1032, an addition processing unit 1070, a virtual processing unit 1040, a control unit 1035, and a user.
  • An I / F 1036 is provided.
  • the input unit 1011 includes an HDMI receiver 1111, a DIR 1112, and an A / D conversion unit 1113.
  • the HDMI receiver 1111 inputs an HDMI signal conforming to the HDMI standard and outputs it to the decoder 1010.
  • the DIR 1112 receives a digital audio signal (SPDIF) and outputs it to the decoder 1010.
  • the A / D conversion unit 1113 receives an analog audio signal, converts the analog audio signal into a digital audio signal, and outputs the digital audio signal to the decoder 1010.
  • the decoder 1010 is a DSP and decodes an input signal.
  • the decoder 1010 inputs signals of various formats such as AAC (registered trademark), Dolby Digital (registered trademark), DTS (registered trademark), MPEG-1 / 2, MPEG-2 multichannel, MP3, etc. It is converted into an audio signal (digital audio signal of FL channel, FR channel, C channel, SL channel, and SR channel. Hereinafter, when simply referred to as an audio signal, the digital audio signal is indicated) and output.
  • a thick solid line shown in FIG. 13 indicates a multi-channel audio signal.
  • the decoder 1010 also has a function of expanding, for example, a stereo channel audio signal to a multi-channel audio signal.
  • the multichannel audio signal output from the decoder 1010 is input to the filter processing unit 1014 and the filter processing unit 1015.
  • the filter processing unit 1014 extracts and outputs a band suitable for each speaker unit from the multi-channel audio signal output from the decoder 1010.
  • FIG. 14A is a block diagram illustrating the configuration of the filter processing unit 1014
  • FIG. 14B is a block diagram illustrating the configuration of the filter processing unit 1015.
  • the filter processing unit 1014 includes HPF1014FL, HPF1014FR, HPF1014C, HPF1014SL, and HPF1014SR that respectively input digital audio signals of the FL channel, FR channel, C channel, SL channel, and SR channel.
  • the filter processing unit 1014 includes an LPF 1015FL, an LPF 1015FR, an LPF 1015C, an LPF 1015SL, and an LPF 1015SR for inputting digital audio signals of the FL channel, the FR channel, the C channel, the SL channel, and the SR channel, respectively.
  • HPF1014FL, HPF1014FR, HPF1014C, HPF1014SL, and HPF1014SR extract and output the high frequency of the input audio signal of each channel.
  • the cut-off frequencies of HPF 1014FL, HPF 1014FR, HPF 1014C, HPF 1014SL, and HPF 1014SR are set to match the lower limit (eg, 200 Hz) of the reproduction frequency of speaker units 1021A to 1021P.
  • Output signals of HPF 1014FL, HPF 1014FR, HPF 1014C, HPF 1014SL, and HPF 1014SR are output to beam forming processing section 1020.
  • LPF1015FL, LPF1015FR, LPF1015C, LPF1015SL, and LPF1015SR each extract and output the low frequency (for example, less than 200 Hz) of the input audio signal of each channel.
  • the cutoff frequencies of LPF1015FL, LPF1015FR, LPF1015C, LPF1015SL, and LPF1015SR correspond to the cutoff frequencies of the HPF1014FL, HPF1014FR, HPF1014C, HPF1014SL, and HPF1014SR (for example, 200 Hz).
  • the output signals of LPF1015FL, LPF1015C, and LPF1015SL are added by an adder 1016 to become an L channel audio signal.
  • the L channel audio signal is further input to the HPF 1030L and the LPF 1031L.
  • HPF1030L extracts and outputs the high frequency of the input audio signal.
  • the LPF 1031L extracts and outputs the low frequency range of the input audio signal.
  • the cutoff frequencies of the HPF 1030L and the LPF 1031L correspond to the crossover frequency (for example, 100 Hz) between the woofer 1033L and the subwoofer 1003.
  • the crossover frequency may be changed by the listener via the user I / F 1036.
  • the output signals of LPF1015FR, LPF1015C, and LPF1015SR are added by the adder 17 to become an R channel audio signal.
  • the R channel audio signal is further input to the HPF 1030R and the LPF 1031R.
  • HPF1030R extracts and outputs the high frequency of the input audio signal.
  • the LPF 1031R extracts and outputs the low frequency range of the input audio signal.
  • the cutoff frequency of the HPF 1030R corresponds to the crossover frequency (for example, 100 Hz) between the woofer 1033R and the subwoofer 1003. As described above, the crossover frequency may be changed by the listener via the user I / F 1036.
  • the audio signal output from the HPF 1030L is input to the woofer 1033L via the addition processing unit 1032.
  • the audio signal output from the HPF 1030R is input to the woofer 1033R via the addition processing unit 1032.
  • the audio signal output from the LPF 1031L and the audio signal output from the LPF 1031R are added to a monaural signal by the addition processing unit 1070 and input to the subwoofer 1003.
  • the addition processing unit 1070 also receives the LFE channel, adds the audio signal output from the LPF 1031L and the audio signal output from the LPF 1031R, and outputs the result to the subwoofer 1003.
  • the filter processing unit 1015 includes an HPF 1040FL, an HPF 1040FR, an HPF 1040C, an HPF 1040SL, and an HPF 1040SR that input digital audio signals of the FL channel, the FR channel, the C channel, the SL channel, and the SR channel, respectively.
  • the filter processing unit 1015 includes an LPF 1041FL, an LPF 1041FR, an LPF 1041C, an LPF 1041SL, and an LPF 1041SR that respectively input digital audio signals of the FL channel, the FR channel, the C channel, the SL channel, and the SR channel.
  • HPF1040FL, HPF1040FR, HPF1040C, HPF1040SL, and HPF1040SR each extract and output the high frequency of the input audio signal of each channel.
  • the cutoff frequencies of HPF 1040FL, HPF 1040FR, HPF 1040C, HPF 1040SL, and HPF 1040SR correspond to the crossover frequency (for example, 100 Hz) between the woofer 1033R and the woofer 1033L and the subwoofer 1003.
  • the crossover frequency may be changed by the listener via the user I / F 1036.
  • the cutoff frequencies of HPF 1040FL, HPF 1040FR, HPF 1040C, HPF 1040SL, and HPF 1040SR may be the same as the cutoff frequencies of HPF 1014FL, HPF 1014FR, HPF 1014C, HPF 1014SL, and HPF 1014SR.
  • the filter processing unit 1015 as a mode including only the HPF 1040FL, HPF 1040FR, HPF 1040C, HPF 1040SL, and HPF 1040SR, the low frequency band may not be output to the subwoofer 1003. Audio signals output from HPF 1040FL, HPF 1040FR, HPF 1040C, HPF 1040SL, and HPF 1040SR are output to virtual processing unit 1040.
  • LPF1041FL, LPF1041FR, LPF1041C, LPF1041SL, and LPF1041SR extract and output the low frequency range of the input audio signal of each channel.
  • the cutoff frequency of the LPF 1041FL, LPF 1041FR, LPF 1041C, LPF 1041SL, and LPF 1041SR corresponds to the crossover frequency (for example, 100 Hz).
  • the audio signals output from the LPF 1041FL, LPF 1041FR, LPF 1041C, LPF 1041SL, and LPF 1041SR are added to a monaural signal by the adder 171 and then input to the subwoofer 1003 via the addition processing unit 1070.
  • the addition processing unit 1070 adds the audio signal output from the LPF 1041FL, LPF 1041FR, LPF 1041C, LPF 1041SL, and LPF 1041SR, the audio signal output from the LPF 1031R and LPF 1031L, and the audio signal of the LFE channel described above.
  • the addition processing unit 1070 may include a gain adjustment unit that changes the addition ratio of these signals.
  • FIG. 15 is a block diagram showing the configuration of the beamization processing unit 1020.
  • the beamization processing unit 1020 includes a gain adjustment unit 1018FL, a gain adjustment unit 1018FR, a gain adjustment unit 1018C, a gain adjustment unit 1018SL, which respectively input digital audio signals of FL channel, FR channel, C channel, SL channel, and SR channel. And a gain adjusting unit 1018SR.
  • the gain adjusting unit 1018FL, the gain adjusting unit 1018FR, the gain adjusting unit 1018C, the gain adjusting unit 1018SL, and the gain adjusting unit 1018SR adjust the gain of the audio signal of each channel.
  • the gain-adjusted audio signals of each channel are input to directivity control unit 1091FL, directivity control unit 1091FR, directivity control unit 1091C, directivity control unit 1091SL, and directivity control unit 1091SR, respectively.
  • the directivity control unit 1091FL, the directivity control unit 1091FR, the directivity control unit 1091C, the directivity control unit 1091SL, and the directivity control unit 1091SR distribute the audio signal of each channel to the speaker units 1021A to 1021P.
  • the distributed audio signals for the speaker units 1021A to 1021P are combined by the combining unit 1092 and supplied to the speaker units 1021A to 1021P.
  • the directivity control unit 1091FL, the directivity control unit 1091FR, the directivity control unit 1091C, the directivity control unit 1091SL, and the directivity control unit 1091SR adjust the delay amount of the audio signal supplied to each speaker unit.
  • the sounds output from the speaker units 1021A to 1021P are strengthened with respect to each other at the same phase and are output as a sound beam having directivity. For example, when sound is output from all speakers at the same timing, a sound beam having directivity is output in front of the array speaker device 1002.
  • the directivity control unit 1091FL, the directivity control unit 1091FR, the directivity control unit 1091C, the directivity control unit 1091SL, and the directivity control unit 1091SR output the sound beam by changing the delay amount to be given to each audio signal. The direction can be changed.
  • directivity control unit 1091FL, directivity control unit 1091FR, directivity control unit 1091C, directivity control unit 1091SL, and directivity control unit 1091SR are such that the phase of each sound output from the speaker units 1021A to 1021P is at a predetermined position. It is also possible to provide a delay amount so that the sound beam is focused and to form an audio beam that focuses at the predetermined position.
  • the sound beam can reach the listening position directly from the array speaker device 1002 or reflected on an indoor wall or the like.
  • the sound beam of the C channel audio signal can be output in the front direction so that the C channel sound beam can reach from the front of the listening position.
  • the sound beams of the FL channel audio signal and the FR channel audio signal are output in the left-right direction of the array speaker device 1002 and reflected on the walls present on the left and right of the listening position, respectively, from the left and right directions of the listening position. Can be reached.
  • the sound beams of the SL channel audio signal and the SR channel audio signal are output in the left-right direction and reflected twice on the wall existing on the left and right of the listening position and on the wall existing on the rear side, respectively. It can be reached from the right rear direction.
  • Such setting of the output direction of the sound beam can be automatically performed by measuring the listening environment using the microphone 1007.
  • the control unit 1035 An audio beam composed of a test signal (for example, white noise) is output to the beam processing unit 1020.
  • the control unit 1035 passes from the left direction parallel to the front surface of the array speaker device 1002 (referred to as a -90 degree direction) to the direction perpendicular to the front surface of the array speaker device 1002 (referred to as a 0 degree direction).
  • the sound beam is turned to the right direction (referred to as a 90-degree direction) parallel to the front surface of 1002.
  • the sound beam is reflected on the wall of the room R according to the turning angle ⁇ of the sound beam, and the sound beam is picked up by the microphone 1007 at a predetermined angle. .
  • the control unit 1035 stores the level of the audio signal input from the microphone 1007 in a memory (not shown) in association with the output angle of the audio beam. Then, the control unit 1035 assigns each channel of the multi-channel audio signal and the output angle of the sound beam based on the peak component of the level of the audio signal. For example, the control unit 1035 detects a peak that is greater than or equal to a predetermined threshold in the collected sound data. The control unit 1035 assigns the output angle of the sound beam at the highest level among these peaks as the output angle of the C channel sound beam. For example, in FIG. 16B, the angle ⁇ 3a at the highest level is assigned as the output angle of the C channel audio beam.
  • control unit 1035 assigns the output angles of the sound beams of the SL channel and the SR channel for the peaks that are close to each other with the peak set for the C channel. For example, in FIG. 16B, an angle ⁇ 2a close to the C channel and close to the ⁇ 90 degree direction is assigned as the SL channel sound beam, and an angle ⁇ 4a close to the C channel and close to the 90 degree direction is assigned to the SR channel sound. Assign as the beam output angle. Further, the control unit 1035 assigns the output angles of the sound beams of the FL channel and the FR channel for the outermost peak. For example, in the example of FIG.
  • the angle ⁇ 1a closest to the ⁇ 90 ° direction is assigned as the FL channel audio beam
  • the angle ⁇ 5a closest to the 90 ° direction is assigned as the output angle of the FR channel audio beam.
  • the control unit 1035 detects the level at which the sound beam of each channel reaches the listening position, and the beam angle that sets the output angle of the sound beam based on the peak of the level measured by the detection unit. Implement setting means.
  • setting is performed so that the sound beam reaches from the surroundings at the position of the listener (microphone 1007).
  • FIG. 17 is a block diagram illustrating a configuration of the virtual processing unit 1040.
  • the virtual processing unit 1040 includes a level adjustment unit 1043, a localization adding unit 1042, a correction unit 1051, a delay processing unit 1060L, and a delay processing unit 1060R.
  • the level adjustment unit 1043 includes a gain adjustment unit 1043FL, a gain adjustment unit 1043FR, a gain adjustment unit 1043C, a gain adjustment unit 1043SL, which respectively input digital audio signals of the FL channel, the FR channel, the C channel, the SL channel, and the SR channel.
  • a gain adjustment unit 1043SR is provided.
  • the gain adjusting unit 1043FL, the gain adjusting unit 1043FR, the gain adjusting unit 1043C, the gain adjusting unit 1043SL, and the gain adjusting unit 1043SR adjust the gain of the audio signal of each channel.
  • the gain of each gain adjustment unit is set by the control unit 1035 based on the test sound beam detection result, for example. For example, as shown in FIG. 16B, the sound beam of the C channel is the highest level because it is a direct sound. Therefore, the gain of the gain adjusting unit 1043C is set to the lowest. Further, the C-channel sound beam is a direct sound and is less likely to depend on the room environment, so may be a fixed value, for example. In the other gain adjustment unit, a gain corresponding to the level difference from the C channel is set.
  • the control unit 1035 rotates the sound beam of the test signal so that the sound beam of each channel reaches the listening position.
  • the listener uses the user I / F 1036 to manually instruct the control unit 1035 to output an audio beam
  • the gain adjustment unit 1043FL, the gain adjustment unit 1043FR, the gain adjustment unit 1043C, The levels of the gain adjustment unit 1043SL and the gain adjustment unit 1043SR may be set manually.
  • the gain adjustment unit 1043FL, gain adjustment unit 1043FR, gain adjustment unit 1043C, gain adjustment unit 1043SL, and gain adjustment unit 1043SR are set for each channel separately from the level detected by sweeping the test audio beam. May be measured.
  • the test sound beam can be output for each channel in the direction determined by the sweep of the test sound beam, and the sound collected by the microphone 1007 at the listening position can be analyzed.
  • the gain-adjusted audio signal of each channel is input to the localization adding unit 1042.
  • the localization adding unit 1042 performs processing for localizing the input audio signal of each channel as a virtual sound source at a predetermined position.
  • a head-related transfer function hereinafter referred to as HRTF
  • HRTF head-related transfer function
  • HRTF is an impulse response that expresses the loudness, arrival time, frequency characteristics, etc. from the virtual speaker installed at a certain position to the left and right ears.
  • the localization adding unit 1042 can cause the listener to localize the virtual sound source by adding HRTF to the input audio signal of each channel and emitting sound from the woofer 1033L or the woofer 1033R.
  • FIG. 18A is a block diagram illustrating a configuration of the localization adding unit 1042.
  • the localization adding unit 1042 includes an FL filter 1421L, an FR filter 1422L, a C filter 1423L, an SL filter 1424L, and an SR filter 1425L for convolving an HRTF impulse response with the audio signal of each channel, an FL filter 1421R, an FR filter 1422R, A C filter 1423R, an SL filter 1424R, and an SR filter 1425R.
  • the audio signal of the FL channel is input to the FL filter 1421L and the FL filter 1421R.
  • the FL filter 1421L adds an HRTF of a path from the position of the virtual sound source VSFL (see FIG. 19A) in front of the listener to the left ear to the FL channel audio signal.
  • the FL filter 1421R adds an HRTF of a path from the position of the virtual sound source VSFL to the right ear to the audio signal of the FL channel.
  • an HRTF from the position of the virtual sound source around the listener to each ear is given.
  • the adder 1426L synthesizes the audio signals to which the HRTF has been assigned by the FL filter 1421L, the FR filter 1422L, the C filter 1423L, the SL filter 1424L, and the SR filter 1425L, and outputs the synthesized audio signal to the correcting unit 1051.
  • Adder 1426R synthesizes the audio signals to which HRTF has been assigned by FL filter 1421R, FR filter 1422R, C filter 1423R, SL filter 1424R, and SR filter 1425R, and outputs the result to audio correction unit 1051 as audio signal VR.
  • the correction unit 1051 performs a crosstalk cancellation process.
  • FIG. 18B is a block diagram illustrating a configuration of the correction unit 1051.
  • the correction unit 1051 includes a direct correction unit 1511L, a direct correction unit 1511R, a cross correction unit 1512L, and a cross correction unit 1512R.
  • the audio signal VL is input to the direct correction unit 1511L and the cross correction unit 1512L.
  • the audio signal VR is input to the direct correction unit 1511R and the cross correction unit 1512R.
  • the direct correction unit 1511L performs processing to make the listener perceive that the sound output from the woofer 1033L is emitted near the left ear.
  • the direct correction unit 1511L is set with a filter coefficient such that the frequency characteristic of the sound output from the woofer 1033L is flat at the position of the left ear.
  • the direct correction unit 1511L processes the input audio signal VL with the filter and outputs an audio signal VLD.
  • filter coefficients are set such that the frequency characteristics of the sound output from the woofer 1033R are flat at the position of the right ear.
  • the direct correction unit 1511R processes the input audio signal VL with the filter and outputs an audio signal VRD.
  • the cross correction unit 1512L is set with a filter coefficient for imparting a frequency characteristic of sound that circulates from the woofer 1033L to the right ear.
  • the sound (VLC) that circulates from the woofer 1033L to the right ear is reversed by the synthesizing unit 1052R and emitted from the woofer 1033R, thereby suppressing the sound of the woofer 1033L from being heard by the right ear. Thereby, the listener perceives the sound emitted from the woofer 1033R as if it was emitted near the right ear.
  • the cross correction unit 1512R is set with a filter coefficient for providing a frequency characteristic of a sound that circulates from the woofer 1033R to the left ear.
  • the sound (VRC) that circulates from the woofer 1033R to the left ear is reversed in phase by the synthesizing unit 1052L and emitted from the woofer 1033L, thereby suppressing the sound of the woofer 1033R from being heard by the left ear. This makes the listener perceive that the sound emitted from the woofer 1033L is emitted near the left ear.
  • the audio signal output from the synthesis unit 1052L is input to the delay processing unit 1060L.
  • the audio signal delayed by a predetermined time by the delay processing unit 1060L is input to the addition processing unit 1032.
  • the audio signal output from the synthesis unit 1052R is input to the delay processing unit 1060R.
  • the audio signal delayed by a predetermined time by the delay processing unit 1060R is input to the addition processing unit 1032.
  • the delay time by the delay processing unit 1060L and the delay processing unit 1060R is set longer than the longest delay time among the delay times given by the directivity control unit of the beam processing unit 1020, for example. Thereby, the sound that perceives the virtual sound source does not hinder the formation of the sound beam. Note that a mode in which a delay processing unit is provided at the subsequent stage of the beam forming processing unit 1020 and a delay is added to the sound beam side so that the sound that causes the sound beam to localize the virtual sound image may be prevented.
  • the audio signal output from the delay processing unit 1060L is input to the woofer 1033L via the addition processing unit 1032.
  • the addition processing unit 1032 adds the audio signal output from the delay processing unit 1060L and the audio signal output from the HPF 1030L.
  • the addition processing unit 1032 may include a gain adjustment unit that changes the addition ratio of these audio signals.
  • the audio signal output from the delay processing unit 1060R is input to the woofer 1033R via the addition processing unit 1032.
  • the addition processing unit 1032 adds the audio signal output from the delay processing unit 1060R and the audio signal output from the HPF 1030R.
  • the addition processing unit 1032 may include a configuration of a gain adjustment unit that changes the addition ratio of these audio signals.
  • a solid arrow indicates a path of an audio beam output from the array speaker apparatus 1002.
  • a white star indicates the position of the sound source generated by the sound beam
  • a black star indicates the position of the virtual sound source.
  • the array speaker device 1002 outputs five sound beams.
  • an audio beam that is focused on a position behind the array speaker device 1002 is set.
  • the listener perceives that the sound source SC is in front of the listener.
  • an audio beam focused on the position of the left front wall of the room R is set, and the listener perceives that the sound source SFL is on the left front wall of the listener.
  • the audio signal of the FR channel is set with a sound beam that focuses on the position of the right front wall of the room R, and the listener perceives that the sound source SFR is on the right front wall of the listener.
  • the audio signal of the SL channel is set with an audio beam focused on the position of the left rear wall of the room R, and the listener perceives that the sound source SSL is on the left rear wall of the listener.
  • the audio signal of the SR channel is set with a sound beam that focuses on the position of the right rear wall of the room R, and the listener perceives that the sound source SSR is on the left rear wall of the listener.
  • the localization adding unit 1042 sets between the C channel sound beam and the FR channel sound beam.
  • the localization adding unit 1042 sets the direction of the virtual sound source VSFR to be symmetric with respect to the arrival direction of the sound beam of the FL channel (symmetric with respect to the listening position as the central axis). This setting may be manually set by the listener using the user I / F 1036, but can be automatically set as follows, for example.
  • control unit 1035 determines the symmetry of the peaks existing on both sides of the peak angle ⁇ a3 set for the C channel.
  • the control unit 1035 determines that the arrival directions of the sound beams of the SL channel and the SR channel are symmetrical if, for example, the allowable error is ⁇ 10 degrees and ⁇ 10 degrees ⁇ ⁇ a2 + ⁇ a4 ⁇ 10 degrees. Similarly, if -10 degrees ⁇ ⁇ a1 + ⁇ a5 ⁇ 10 degrees, the control unit 1035 determines that the sound beam arrival directions of the FL channel and the FR channel are symmetrical.
  • FIG. 19B shows an example in which the value of ⁇ a1 + ⁇ a5 exceeds the allowable error. Therefore, the control unit 1035 instructs the localization adding unit 1042 to set the direction of the virtual sound source between the arrival directions of the two sound beams (the C channel sound beam and the FR channel sound beam). To do.
  • the direction of the virtual sound source is preferably set so as to be symmetric with the sound beam on the side close to the ideal arrival direction (for example, about 30 degrees left and right as viewed from the listening position).
  • the positions of the virtual sound sources are set at substantially the same positions as the positions of the sound sources SFL, SC, SSL, and SSR. Therefore, the listener perceives the virtual sound sources VSC, VSFL, VSSL, and VSSR at substantially the same positions as the positions of the sound sources SC, SFL, SSL, and SSR.
  • the array speaker apparatus 1002 clearly locates the sound source in the intended direction using the virtual sound source based on the head-related transfer function that does not depend on the listening environment such as the acoustic reflectivity of the wall, while using the sense of localization by the sound beam. be able to. Further, in the examples of FIGS. 19A and 19B, the sound source is localized in a bilaterally symmetric direction when viewed from the listening position, so that a more ideal listening mode is obtained.
  • FIG. 20A is a diagram illustrating an example in which the SR channel reaches closer to the front than the SL channel.
  • the distance between the right wall and the listening position is longer than the distance between the left wall and the listening position. Since the surround channel is reflected twice, the sound source SSR is perceived closer to the front than the sound source SSL when the right wall surface is far.
  • the control unit 1035 determines whether or not ⁇ 10 degrees ⁇ ⁇ a2 + ⁇ a4 ⁇ 10 degrees, for example, with an allowable error of ⁇ 10 degrees.
  • FIG. 20B shows an example in which the value of ⁇ a2 + ⁇ a4 exceeds the allowable error. Therefore, the control unit 1035 instructs the localization adding unit 1042 to set the direction of the virtual sound source between the arrival directions of the two sound beams.
  • the direction of the virtual sound source is preferably set so as to be symmetric with the sound beam on the side close to the ideal arrival direction (for example, about 110 degrees left and right as viewed from the listening position).
  • the surround channel has an ideal arrival direction closer to the front left / right direction than the front channel, so the direction of the virtual sound source is set to the peak (audio beam reaching the left / right side) with a large angle difference from the central axis.
  • the direction of the virtual sound source VSSL is set to ⁇ a2 ′ that is symmetrical to ⁇ a4 across the central axis (here, ⁇ a3).
  • the positions of the virtual sound sources are set at substantially the same positions as the positions of the sound sources SFL, SFR, SC, and SSR. Therefore, the listener perceives the virtual sound sources VSC, VSFR, VSSL, and VSSR at substantially the same positions as the positions of the sound sources SC, SFR, SSL, and SSR.
  • the sound source is localized in a symmetric direction when viewed from the listening position for the surround channel, which is a more ideal listening mode.
  • the array speaker device 1002 can supplement the sense of localization with the virtual sound source VSSL and the virtual sound source VSSR that are generated by the sound directly reaching the ears of the listener by the woofer 1033L and the woofer 1033R, and can clearly hear the sound source in a more ideal direction. Can be localized.
  • FIG. 21 is a block diagram showing a configuration of the array speaker device 1002A when the phantom sound source is used together.
  • the components common to the array speaker device 1002 in FIG. 13 are denoted by the same reference numerals and description thereof is omitted.
  • the array speaker device 1002A is different from the array speaker device 1002 in that a phantom processing unit 1090 is provided.
  • the phantom processing unit 1090 distributes the audio signal of each channel among the audio signals input from the filter processing unit 1014 to its own channel and other channels, thereby phantom localization of a specific channel (generates a phantom sound source) )
  • FIG. 22A is a block diagram illustrating a configuration of the phantom processing unit 1090.
  • FIG. 22B is a diagram illustrating a correspondence table between designated angles and gain ratios.
  • FIG. 22C is a diagram illustrating a correspondence table between the specified angle and the filter coefficient (the head related transfer function provided by the localization adding unit 1042).
  • the phantom processing unit 1090 includes a gain adjustment unit 1095FL, a gain adjustment unit 1096FL, a gain adjustment unit 1095FR, a gain adjustment unit 1096FR, a gain adjustment unit 1095SL, a gain adjustment unit 1096SL, a gain adjustment unit 1095SR, a gain adjustment unit 1096SR, an addition unit 1900, An adder 1901 and an adder 1902 are provided.
  • the audio signal of the FL channel is input to the gain adjustment unit 1095FL and the gain adjustment unit 1096FL.
  • the FR channel audio signal is input to the gain adjustment unit 1095FR and the gain adjustment unit 1096FR.
  • SL channel audio signals are input to gain adjustment section 1095SL and gain adjustment section 1096SL.
  • An SR channel audio signal is input to gain adjustment section 1095SR and gain adjustment section 1096SR.
  • the gain ratio of the audio signal of the FL channel is adjusted by the gain adjusting unit 1095FL and the gain adjusting unit 1096FL and input to the adding unit 1901 and the adding unit 1900, respectively.
  • the gain ratio of the audio signal of the FR channel is adjusted by the gain adjustment unit 1095FR and the gain adjustment unit 1096FR, and is input to the addition unit 1902 and the addition unit 1900, respectively.
  • the gain ratio of the audio signal of the SL channel is adjusted by the gain adjustment unit 1095SL and the gain adjustment unit 1096SL, and is input to the beam forming processing unit 1020 and the addition unit 1901, respectively.
  • the gain ratio of the audio signal of the SR channel is adjusted by the gain adjustment unit 1095SR and the gain adjustment unit 1096SR, and is input to the beam processing unit 1020 and the addition unit 1902, respectively.
  • the gain of each gain adjustment unit is set by the control unit 1035.
  • the control unit 1035 reads the association table stored in the memory (not shown), and reads the gain ratio associated with the designated angle.
  • the control unit 1035 controls the gain ratio between the FR channel sound beam arriving from the front right of the listening position and the C channel sound beam arriving from the front of the listening position. Control the direction of the phantom sound source.
  • the controller 1035 Since the designated angle is 40 degrees, the controller 1035 has an FR channel sound beam arrival direction ⁇ a5 (FR angle) of 80 degrees, and a C channel sound beam arrival direction ⁇ a3 (C angle) of 0 degrees.
  • the control unit 1035 sets the gain of the gain adjustment unit 1095FR to 0.5 and the gain of the gain adjustment unit 1096FR to 0.5.
  • the phantom sound source can be localized in the direction of 40 degrees to the right between the sound beam of the FR channel and the sound beam of the C channel reaching from the front of the listening position.
  • the gain may be set so that the power is constant.
  • the gain of the gain adjustment unit 1095FR and the gain of the gain adjustment unit 1096FR are ⁇ 3 dB (about 0.707).
  • control unit 1035 reads out the filter coefficient for localization of the virtual sound source in the direction of 40 degrees that is the specified angle from the table shown in FIG. 22C, and sets it in the localization adding unit 1042. Thereby, the virtual sound source VSFR is localized in the same direction as the phantom sound source SFR.
  • the designated angle can be manually input by the listener via the user I / F 1036, but can also be automatically set using the measurement result of the test sound beam described above.
  • the arrival direction ⁇ a1 of the FL channel audio beam is ⁇ 60 degrees (60 degrees left when viewed from the listening position), and the FR phantom sound source is localized in a direction symmetrical to the arrival direction of the FL channel audio beam.
  • the array speaker apparatus 1002A supplements the sense of localization of the phantom sound source by the sound beam with the virtual sound source by the head related transfer function that does not depend on the listening environment such as the acoustic reflectivity of the wall, and more clearly locates the phantom sound source. be able to.
  • the array speaker device 1002A can supplement the sense of localization with the virtual sound source VSSL and the virtual sound source VSSR that are generated by the sound directly reaching the ears of the listener by the woofer 1033L and the woofer 1033R, and can localize the phantom sound source more clearly. Can do.
  • FIG. 24 is a diagram illustrating an example in which an audio signal of 7.1 channel is localized using five sound beams.
  • 7.1 channel surround includes 2 channels (SBL, SBR) to be played from behind the listener.
  • the array speaker apparatus 1002A sets the SBL channel to an audio beam that focuses on the position of the left rear wall of the room R, and the audio beam that focuses the SBR channel on the position of the right rear left wall of the room R. Set to.
  • the array speaker apparatus 1002A uses the SBL channel and FL channel sound beams to set the SL channel phantom sound source SSL at a position between them (-90 degrees to the left of the listening position).
  • the phantom sound source SSR of the SR channel is set at a position between the sound beams of the SBR channel and the FR channel (90 degrees to the right of the listening position) between them.
  • the array speaker apparatus 1002A sets the virtual sound source VSSL at the position of the phantom sound source SSL and sets the virtual sound source VSSR at the position of the phantom sound source SSR.
  • the array speaker device 1002A can supplement the sense of localization with a virtual sound source generated by sounds that reach the listener's ear directly by the woofer 1033L and the woofer 1033R. A large number of channels can be localized more clearly.
  • FIG. 25 (A) is a diagram showing an array speaker device 1002B according to a modification. A description of the same configuration as that of the array speaker device 1002 is omitted.
  • Array speaker device 1002B is different from array speaker device 1002 in that sounds output from woofer 1033L and woofer 1033R are output from speaker unit 1021A and speaker unit 1021P, respectively.
  • the array speaker apparatus 1002B outputs sound that makes a virtual sound source perceived from the speaker units 1021A and 1021P at both ends of the speaker units 1021A to 1021P.
  • the speaker unit 1021A and the speaker unit 1021P are the speaker units arranged at the extreme end of the array speaker, and are arranged on the left and right sides as viewed from the listener. Therefore, the speaker unit 1021A and the speaker unit 1021P are suitable for outputting the sound of the L channel and the R channel, respectively, and are suitable as the speaker unit for outputting the sound for causing the virtual sound source to be perceived.
  • the array speaker apparatus 1002 does not have to include all the speaker units 1021A to 1021P, the woofer 1033L, and the woofer 1033R in one housing.
  • each speaker unit may be provided in a separate housing, and the housings may be arranged side by side.
  • FIG. 26 is a diagram for explaining an AV system 2001 provided with an array speaker device 2002.
  • FIG. 27 is a part of a block diagram of the array speaker device 2002 and the subwoofer 2003.
  • FIG. 28A is a block diagram of the initial reflected sound processing unit 2022
  • FIG. 28B is a block diagram of the rear reflected sound processing unit 2044.
  • FIG. 29 is a schematic diagram showing an example of an impulse response actually measured in a concert hall.
  • 30A is a block diagram of the localization adding unit 2042
  • FIG. 30B is a block diagram of the correcting unit 2051.
  • FIG. 31 is a diagram for explaining the sound output by the array speaker apparatus 2002.
  • FIG. 31 is a diagram for explaining the sound output by the array speaker apparatus 2002.
  • the AV system 2001 includes an array speaker device 2002, a subwoofer 2003, and a television 2004.
  • the array speaker device 2002 is connected to the subwoofer 2003 and the television 2004.
  • the array speaker apparatus 2002 receives an audio signal corresponding to a video reproduced on the television 2004 or a content audio signal from a content player (not shown).
  • the array speaker device 2002 outputs sound having a directivity and a virtual sound source perceived based on the audio signal of the input content, and further gives a sound field effect to the sound of the content.
  • the array speaker device 2002 includes, for example, a rectangular parallelepiped housing.
  • the housing of the array speaker apparatus 2002 includes, for example, 16 speaker units 2021A to 2021P and woofers 2033L and 2033R (corresponding to the first sound emission unit of the present invention) on the surface facing the listener.
  • the number of speaker units is not limited to 16, and may be, for example, 8 or the like.
  • Speaker units 2021A to 2021P are arranged in a row.
  • the speaker units 2021A to 2021P are arranged in order from the left side when the array speaker device 2002 is viewed from the listener.
  • Woofer 2033L is arranged further to the left from speaker unit 2021A.
  • the woofer 2033R is arranged on the right side of the speaker unit 2021P.
  • the array speaker apparatus 2002 includes a decoder 2010 and a directivity control unit 2020 as shown in FIG. Note that the set of the speaker units 2021A to 2021P and the directivity control unit 2020 corresponds to the second sound emitting unit of the present invention.
  • the decoder 2010 is connected to a DIR (; Digital audio I / F Receiver) 2011, ADC (; Analog to Digital Converter) 2012, and HDMI (registered trademark; High Definition Multimedia Interface) receiver 13.
  • DIR Digital audio I / F Receiver
  • ADC Analog to Digital Converter
  • HDMI registered trademark; High Definition Multimedia Interface
  • the DIR2011 receives a digital audio signal transmitted through an optical cable or a coaxial cable.
  • the ADC 2012 converts the input analog signal into a digital signal.
  • the HDMI receiver 2013 receives an HDMI signal that conforms to the HDMI standard.
  • the decoder 2010 supports various data formats such as AAC (registered trademark), Dolby Digital (registered trademark), DTS (registered trademark), MPEG-1 / 2, MPEG-2 multichannel, or MP3.
  • the decoder 2010 converts the digital audio signals output from the DIR 2011 and the ADC 2012 into multi-channel audio signals (FL channel, FR channel, C channel, SL channel, and SR channel digital audio signals.
  • the audio signals are simply referred to as audio signals).
  • a digital audio signal The decoder 2010 extracts audio data from the HDMI signal (a signal conforming to the HDMI standard) output from the HDMI receiver 2013, decodes the audio data, and outputs the audio signal.
  • the decoder 2010 can convert the audio signal into various audio signals such as a 7-channel audio signal as well as a 5-channel audio signal.
  • the array speaker apparatus 2002 divides the band of the audio signal output from the decoder 2010, outputs a high frequency (for example, 200 Hz or more) to the speaker units 2021A to 2021P, and outputs a low frequency (for example, less than 200 Hz) to the woofers 2033L and 2033R.
  • HPF 2014 (2014FL, 2014FR, 2014C, 2014SR, 2014SL)
  • LPF2015 (2015FL, 2015FR, 2015C, 2015SR, 2015SL) are provided for output to subwoofer unit 2072.
  • the cutoff frequencies of the HPF 2014 and the LPF 2015 are set so as to match the lower limit (200 Hz) of the reproduction frequency of the speaker units 2021A to 2021P.
  • the audio signal of each channel output from the decoder 2010 is input to the HPF 2014 and the LPF 2015, respectively.
  • the HPF 2014 extracts and outputs a high frequency component (200 Hz or more) of the input audio signal.
  • the LPF 2015 extracts and outputs a low frequency component (less than 200 Hz) of the input audio signal.
  • the array speaker apparatus 2002 includes an initial reflected sound processing unit 2022 in order to give a sound field effect of the initial reflected sound to the sound of the content.
  • Each audio signal output from the HPF 2014 is input to the initial reflected sound processing unit 2022.
  • the initial reflected sound processing unit 2022 superimposes the audio signal of the initial reflected sound on each input audio signal and outputs the superimposed audio signal to the level adjustment unit 2018 (2018FL, 2018FR, 2018C, 2018SR, 2018SL).
  • the initial reflected sound processing unit 2022 includes a gain adjusting unit 2221, an initial reflected sound generating unit 2222, and a synthesizing unit 2223 as shown in FIG.
  • Each audio signal input to the initial reflected sound processing unit 2022 is input to the gain adjustment unit 2221 and the synthesis unit 2223.
  • the gain adjusting unit 2221 inputs the level of each input audio signal and the gain adjusting unit 2441 (see FIG. 28B) in order to adjust the level ratio between the initial reflected sound and the rear reverberant sound. Then, the level ratio with each audio signal level is adjusted, and each audio signal after level adjustment is output to the initial reflected sound generation unit 2222.
  • the initial reflected sound generation unit 2222 generates an audio signal of the initial reflected sound based on each input audio signal.
  • the audio signal of the initial reflected sound is generated so as to reflect the actual arrival direction of the initial reflected sound and the delay time of the initial reflected sound.
  • the actual initial reflected sound is generated until a predetermined time (for example, within 300 msec) elapses from the generation of the direct sound (time 0 in the schematic diagram of FIG. 29). Since the actual initial reflected sound has a smaller number of reflections than the rear reverberation sound, the reflection pattern differs for each direction of arrival. Therefore, the actual initial reflected sound has different frequency characteristics for each direction of arrival.
  • the audio signal of such early reflection sound is generated by, for example, an FIR filter, and is generated by convolving a predetermined coefficient with the input audio signal.
  • the predetermined coefficient is set based on sampling data of the impulse response of the actual initial reflected sound shown in FIG. 29, for example.
  • the audio signal of the initial reflection sound generated by the initial reflection sound generation unit 2222 is distributed and output to the audio signal of each channel according to the actual arrival direction of the initial reflection sound.
  • the initial reflected sound is generated so as to be generated discretely until a predetermined time (for example, within 300 msec) elapses from the direct sound (corresponding to the audio signal directly input from the HPF 2014 to the synthesizing unit 2223).
  • Each audio signal output from the initial reflected sound generation unit 2222 is input to the synthesis unit 2223.
  • the synthesizer 2223 outputs an audio signal obtained by synthesizing the audio signal input from the HPF 2014 and the audio signal input from the initial reflection sound generator 2222 to the level adjuster 2018 for each channel. Accordingly, the initial reflected sound is superimposed on the direct sound (corresponding to the audio signal directly input from the HPF 2014 to the synthesis unit 2223). In other words, the initial reflected sound characteristic is added to the direct sound. This initial reflected sound is output as a sound beam together with the direct sound.
  • the level adjustment unit 2018 is provided for adjusting the level of the sound beam for each channel.
  • the level adjusting unit 2018 adjusts and outputs the level of each audio signal.
  • the directivity control unit 2020 receives each audio signal output from the level adjustment unit 2018.
  • the directivity control unit 2020 distributes the input audio signals of the respective channels by the number of the speaker units 2021A to 2021P, and delays them by a predetermined delay time.
  • the delayed audio signal of each channel is converted into an analog audio signal by a DAC (Digital to Analog Converter) (not shown) and then input to the speaker units 2021A to 2021P.
  • the speaker units 2021A to 2021P emit sound based on the input audio signal of each channel.
  • each output from the speaker units 2021A to 2021P is performed.
  • the sound strengthens the phase in a direction corresponding to the difference in the delay amount.
  • the sound beam is formed as a parallel wave traveling in a predetermined direction from the speaker units 2021A to 2021P.
  • the directivity control unit 2020 can perform delay control so that the phases of the sounds output from the speaker units 2021A to 2021P are aligned at predetermined positions. In this case, each sound output from the speaker units 2021A to 2021P becomes an audio beam having a focus on the predetermined position.
  • the array speaker apparatus 2002 may include an equalizer for each channel before or after the directivity control unit 2020 to adjust the frequency characteristics of each audio signal.
  • the audio signal output from the LPF 2015 is input to the woofer 2033L or 33R and the subwoofer unit 2072.
  • the array speaker apparatus 2002 divides an audio signal (less than 200 Hz) other than the sound beam band into a band for the woofers 2033L and 2033R (for example, 100 Hz or more) and a band for the subwoofer unit 2072 (for example, less than 100 Hz).
  • HPF 2030 (2030L, 2030R) and LPF 2031 (2031L, 2031R).
  • the cutoff frequencies of the HPF 2030 and the LPF 2031 are set so as to match the upper limit (100 Hz) of the reproduction frequency of the subwoofer unit 2072.
  • the audio signal (less than 200 Hz) output from the LPF 2015 (2015FL, 2015C, 2015SL) is added by the adding unit 16.
  • the audio signal added by the adding unit 16 is input to the HPF 2030L and the LPF 2031L.
  • the HPF 2030L extracts and outputs a high frequency component (100 Hz or more) of the input audio signal.
  • the LPF 2031L extracts and outputs a low frequency component (less than 100 Hz) of the input audio signal.
  • the audio signal output from the HPF 2030L is input to the woofer 2033L via the level adjustment unit 2034L, the addition unit 2032L, and a DAC (not shown).
  • the audio signal output from the LPF 2031L is input to the subwoofer unit 2072 of the subwoofer 2003 via the level adjustment unit 2070F, the addition unit 2071, and the DAC (not illustrated).
  • the level adjustment unit 2034L and the level adjustment unit 2070F adjust the level of the input audio signal in order to adjust the level ratio of the sound beam, the sound output from the woofer 2033L, and the sound output from the subwoofer unit 2072. Adjust and output.
  • the audio signals output from the LPF 2015 are added by the adding unit 17.
  • the audio signal added by the adding unit 17 is input to the HPF 2030R and the LPF 2031R.
  • the HPF 2030R extracts and outputs a high frequency component (100 Hz or more) of the input audio signal.
  • the LPF 2031R extracts and outputs a low frequency component (less than 100 Hz) of the input audio signal.
  • the audio signal output from the HPF 2030R is input to the woofer 2033R via the level adjustment unit 2034R, the addition unit 2032R, and a DAC (not shown).
  • the audio signal output from the LPF 2031R is input to the subwoofer unit 2072 via the level adjustment unit 2070G, the addition unit 2071, and a DAC (not shown).
  • the level adjuster 2034R and the level adjuster 2070G adjust the level of the input audio signal in order to adjust the level ratio of the sound beam, the sound output from the woofer 2033R, and the sound output from the subwoofer unit 2072. Adjust and output.
  • the array speaker apparatus 2002 outputs the sound beam on which the initial reflected sound is superimposed from the speaker units 2021A to 2021P for each channel, and outputs sounds other than the sound beam band (less than 200 Hz) to the woofers 2033L and 2033R. And output from the subwoofer unit 2072.
  • HPF2040FL, HPF2040FR, HPF2040C, HPF2040SL, and HPF2040SR may be the same as the cutoff frequencies of HPF2014FL, HPF2014FR, HPF2014C, HPF2014SL, and HPF2014SR.
  • the low frequency may not be output to the subwoofer 2003.
  • the array speaker apparatus 2002 includes a rear reflected sound processing unit 2044, a localization adding unit 2042, a crosstalk cancellation processing unit 2050, and delay processing units 2060L and 2060R.
  • the array speaker apparatus 2002 divides the band of the audio signal output from the decoder 2010, outputs the high frequency (for example, 100 Hz or more) to the woofers 2033L and 2033R, and outputs the low frequency (for example, less than 100 Hz) to the subwoofer unit 2072.
  • HPF 2040 (2040FL, 2040FR, 2040C, 2040SR, 2040SL) and LPF 2041 (2041FL, 2041FR, 2041C, 2041SR, 2041SL) are provided.
  • the cutoff frequencies of the HPF 2040 and the LPF 2041 are set so as to match the upper limit (100 Hz) of the reproduction frequency of the subwoofer unit 2072.
  • the audio signal of each channel output from the decoder 2010 is input to the HPF 2040 and the LPF 2041, respectively.
  • the HPF 2040 extracts and outputs a high frequency component (100 Hz or more) of the input audio signal.
  • the LPF 2041 extracts and outputs a low frequency component (less than 100 Hz) of the input audio signal.
  • the array speaker apparatus 2002 includes level adjusting units 2070A to 2070E in order to adjust the level ratio between the sound output from the woofers 2033L and 2033R and the sound output from the subwoofer unit 2072.
  • the level of each audio signal output from the LPF 2041 is adjusted by the level adjusters 2070A to 2070E.
  • the audio signals whose levels have been adjusted by the level adjusting units 2070A to 2070E are added by the adding unit 2071, respectively.
  • the audio signal added by the adder 2071 is input to the subwoofer unit 2072 via a DAC (not shown).
  • Each audio signal output from the HPF 2040 is input to the rear reflection sound processing unit 2044.
  • the rear reflection sound processing unit 2044 superimposes the audio signal of the rear reverberation sound on each input audio signal, and outputs it to the level adjustment unit 2043 (2043FL, 2043FR, 2043C, 2043SR, 2043SL).
  • the rear reflection processing unit 2044 includes a gain adjustment unit 2441, a rear reverberation generation unit 2442, and a synthesis unit 2443, as shown in FIG.
  • Each audio signal input to the rear reflection sound processing unit 2044 is input to the gain adjustment unit 2441 and the synthesis unit 2443.
  • the gain adjusting unit 2441 adjusts the level ratio between the initial reflected sound and the rear reverberant sound, and the level of each input audio signal and each audio input to the gain adjusting unit 2221 of the initial reflected sound processing unit 2022.
  • the level ratio with the signal level is adjusted, and each audio signal after level adjustment is output to the rear reverberation sound generation unit 2442.
  • the rear reverberation sound generation unit 2442 generates an audio signal of the rear reverberation sound based on each input audio signal.
  • the actual rear reverberation sound is generated for a predetermined time (for example, 2 seconds) after the initial reflection sound. Since the actual rear reverberation sound has a larger number of reflections than the initial reflection sound, the reflection pattern is substantially uniform regardless of the direction of arrival. Therefore, the frequency components of the rear reverberation sound are substantially the same regardless of the arrival direction.
  • the rear reverberation generator 2442 includes, for each channel, a configuration in which, for example, comb filters and all-pass recursive filters (IIR filters) are combined in multiple stages in order to generate such rear reverberation.
  • the coefficient of each filter is set so as to have the characteristics of the actual rear reverberation sound (the delay time from the direct sound, the length of the rear reverberation sound, and the amount of attenuation with respect to the length of the rear reverberation sound).
  • the rear reverberant sound is generated so as to be generated after the generation time of the initial reflected sound generated by the initial reflected sound generating unit 2222 (300 msec from the direct sound generation) has elapsed.
  • the rear reverberation sound generation unit 2442 generates an audio signal of the rear reverberation sound for each channel, for example, after 300 msec from the direct sound generation and 2,000 msec elapses, and outputs the audio signal to the synthesis unit 2443.
  • generation part 2442 with an IIR filter was shown, it is realizable also using a FIR filter.
  • Each audio signal output from the rear reverberation sound generation unit 2442 is input to the synthesis unit 2443.
  • the synthesis unit 2443 synthesizes the audio signals input from the rear reverberation sound generation unit 2442 with the audio signals input from the HPF 2040, and combines the synthesized audio signals.
  • the signal is output to the level adjustment unit 2043.
  • the rear reverberation sound is superimposed on the direct sound (corresponding to the audio signal directly input from the HPF 2040 to the synthesis unit 2443).
  • the characteristic of the rear reverberant sound is added to the direct sound.
  • the rear reverberation sound is output from the woofers 2033L and 2033R together with the sound that makes the virtual sound source be perceived.
  • the level adjustment unit 2043 adjusts the level of each input audio signal and outputs it to the localization adding unit 2042 in order to adjust the level of the sound that causes the virtual sound source to be perceived for each channel.
  • the localization adding unit 2042 performs processing to localize each input audio signal to a virtual sound source position.
  • a head-related transfer function hereinafter referred to as HRTF
  • HRTF head-related transfer function
  • HRTF is an impulse response that expresses the loudness, arrival time, frequency characteristics, etc. from the virtual speaker installed at a certain position to the left and right ears.
  • the localization adding unit 2042 includes filters 2421L to 2425L and filters 2421R to 2425R for convolving the HRTF impulse response for each channel, as shown in FIG.
  • the FL channel audio signal (the audio signal output from the HPF 2040FL) is input to the filters 2421L and 2421R.
  • the filter 2421L adds an HRTF of a path from the position of the virtual sound source VSFL (see FIG. 31) in front of the listener to the left ear to the FL channel audio signal.
  • the filter 2421R adds an HRTF of a path from the position of the virtual sound source VSFL to the right ear to the FL channel audio signal.
  • the filter 422L adds the HRTF of the route from the position of the virtual sound source VSFR in front of the listener to the left ear of the listener to the audio signal of the FR channel.
  • the filter 422R adds the HRTF of the path from the position of the virtual sound source VSFR to the right ear to the audio signal of the FR channel.
  • Filters 2423L to 2425L add HRTFs in the path from the positions of virtual sound sources VSC, VSSL, and VSSR corresponding to the C, SL, and SR channels to the listener's left ear to the audio signals of the C, SL, and SR channels.
  • the filters 2423R to 2425R add the HRTF of the path from the position of the virtual sound source VSC, VSSL, VSSR corresponding to the C, SL, SR channel to the right ear of the listener to the audio signal of the C, SL, SR channel.
  • the adding unit 2426L synthesizes the audio signals output from the filters 2421L to 2425L, and outputs them to the crosstalk cancellation processing unit 2050 as the audio signal VL.
  • the adder 2426R combines the audio signals output from the filters 2421R to 2425R, and outputs the synthesized audio signal VR to the crosstalk cancellation processing unit 2050.
  • the crosstalk cancellation processing unit 2050 cancels the crosstalk that is emitted from the woofer 2033L and reaches the right ear, and the woofer 2033L and the woofer 2033R so that the direct sound that is emitted from the woofer 2033L and reaches the left ear can be heard flat. Change the frequency characteristics of each input audio signal. Similarly, the crosstalk cancellation processing unit 2050 cancels the crosstalk that is emitted from the woofer 2033R and reaches the left ear, and the direct sound that is emitted from the woofer 2033R and reaches the right ear is heard flatly. The frequency characteristic of each audio signal input to the woofer 2033R is changed.
  • the crosstalk cancellation processing unit 2050 performs processing using the correction unit 2051 and the combining units 2052L and 2052R.
  • the correction unit 2051 includes direct correction units 2511L and 2511R and cross correction units 2512L and 2512R.
  • the audio signal VL is input to the direct correction unit 2511L and the cross correction unit 2512L.
  • the audio signal VR is input to the direct correction unit 2511R and the cross correction unit 2512R.
  • the direct correction unit 2511L performs a process of making the listener perceive that the sound emitted from the woofer 2033L is emitted near the left ear.
  • the direct correction unit 2511L is set with a filter coefficient so that the sound output from the woofer 2033L can be heard flat at the position of the left ear.
  • the direct correction unit 2511L corrects the input audio signal VL and outputs an audio signal VLD.
  • the cross correction unit 2512R in combination with the synthesizing unit 2052L, outputs from the woofer 2033L a sound having a phase opposite to that from the woofer 2033R and cancels the sound pressure at the position of the left ear, thereby canceling the sound of the woofer 2033R. Is suppressed from being heard in the left ear.
  • the cross correction unit 2512R performs a process of making the listener perceive that the sound emitted from the woofer 2033L is emitted near the left ear.
  • the cross correction unit 2512R is set with a filter coefficient that prevents the sound output from the woofer 2033R from being heard at the position of the left ear.
  • the cross correction unit 2512R corrects the input audio signal VR and outputs an audio signal VRC.
  • the synthesizing unit 2052L makes the audio signal VRC out of phase and synthesizes it with the audio signal VLD.
  • the direct correction unit 2511R performs processing to make the listener perceive that the sound emitted from the woofer 2033R is emitted near the right ear.
  • the direct correction unit 2511R is set with a filter coefficient so that the sound output from the woofer 2033R can be heard flat at the position of the right ear.
  • the direct correction unit 2511R corrects the input audio signal VR and outputs an audio signal VRD.
  • the cross correction unit 2512L in combination with the synthesizing unit 2052R, outputs from the woofer 2033R a sound having a phase opposite to that from the woofer 2033L and cancels the sound pressure at the position of the right ear, thereby canceling the sound of the woofer 2033L. Is suppressed from being heard in the right ear.
  • the cross correction unit 2512L performs processing to make the listener perceive that the sound emitted from the woofer 2033R is emitted near the right ear.
  • the cross correction unit 2512L is set with a filter coefficient so that the sound output from the woofer 2033L cannot be heard at the position of the right ear.
  • the cross correction unit 2512L corrects the input audio signal VL and outputs an audio signal VLC.
  • the synthesizing unit 2052R synthesizes the audio signal VLC with the audio signal VRD in reverse phase.
  • the audio signal output from the synthesis unit 2052L is input to the delay processing unit 2060L.
  • the audio signal is delayed by a predetermined time by the delay processing unit 2060L and input to the level adjusting unit 61L.
  • the audio signal output from the synthesis unit 2052R is input to the delay processing unit 2060R.
  • the delay processing unit 2060R delays the audio signal by the same delay time as the delay processing unit 2060L.
  • the delay time by the delay processing units 2060L and 2060R is set so that the sound beam and the sound that perceives the virtual sound source are not output at the same timing. Thereby, the formation of the sound beam is less likely to be hindered by the sound perceived by the virtual sound source.
  • the array speaker apparatus 2002 includes a delay processing unit for each channel in the subsequent stage of the directivity control unit 2020, and delays the sound beam so that the sound beam does not interfere with the sound that perceives the virtual sound source. It does not matter.
  • the level adjustment units 2061L and 2061R are provided to collectively adjust the sound levels that cause the virtual sound sources of all channels to be perceived.
  • the level adjusters 2061L and 2061R adjust the level of each audio signal delayed by the delay processors 2060L and 2060R.
  • the audio signals whose levels are adjusted by the level adjustment units 2061L and 2061R are input to the woofers 2033L and 2033R via the addition units 2032L and 2032R.
  • the audio signals outside the audio beam band (less than 200 Hz) output from the speaker units 2021A to 2021P are input to the adders 2032L and 2032R, the sound outside the audio beam band and the sound that localizes the virtual sound source Are output by the woofers 2033L and 2033R.
  • the array speaker apparatus 2002 localizes the audio signal of each channel on which the audio signal of the rear reverberant sound is superimposed at a virtual sound source position.
  • a white arrow indicates a path of an audio beam output from the array speaker apparatus 2002
  • a plurality of arcs indicate sounds perceiving a virtual sound source output from the array speaker apparatus 2002.
  • a star indicates the position of the sound source generated by the sound beam and the position of the virtual sound source.
  • the array speaker apparatus 2002 outputs five sound beams according to the number of channels of the input audio signal.
  • the C channel audio signal is subjected to delay control such that the focal position is set behind the array speaker apparatus 2002. Then, the listener perceives that the sound source SC of the C channel audio signal is in front of the listener.
  • the audio signals of the FL and FR channels are delay-controlled so that the sound beam is focused on the left front wall of the listener and the right front wall of the listener.
  • Each sound beam based on the audio signals of the FL and FR channels is reflected once by the wall of the room R and reaches the position of the listener. Then, the listener perceives that the sound sources SFL and SFR of the audio signals of the FL and FR channels are on the left front wall and the right front wall of the listener.
  • the audio signals of the SL and SR channels are subjected to delay control, for example, so that the sound beam is directed toward the left side wall and the right side wall of the listener.
  • Each sound beam based on the audio signals of the SL and SR channels is reflected by the wall of the room R and reaches the left rear wall of the listener and the right rear wall of the listener.
  • Each sound beam is reflected again by the left rear wall of the listener and the right rear wall of the listener to reach the position of the listener. Then, the listener perceives that the sound sources VSSL, VSSSR of the audio signals of the SL and SR channels are on the left rear wall of the listener and the right rear wall of the listener.
  • the filters 2421L to 2425L and filters 2421R to 2425R of the localization adding unit 2042 are set so that the positions of the virtual speakers are substantially the same as the positions of the sound sources SFL, SFR, SC, SSL, and SSR, respectively. Then, as shown in FIG. 31, the listener perceives the virtual sound sources VSC, VSFL, VSFR, VSSL, VSSR at substantially the same positions as the positions of the sound sources SFL, SFR, SC, SSL, SSR.
  • the array speaker apparatus 2002 improves the sense of localization as compared with the case where only the sound beam is used or the case where only the virtual sound source is used.
  • the array speaker apparatus 2002 superimposes the initial reflected sound on each sound beam as shown in FIG.
  • the sound that perceives the virtual sound source is not superposed with the initial reflected sound having different frequency characteristics for each direction of arrival, so that the frequency characteristics of the head-related transfer function are maintained.
  • the sound that makes the virtual sound source perceive has a sense of localization due to the difference in frequency characteristics between both ears, the arrival time difference of the sound, and the volume difference, so the rear reverberation sound with uniform frequency characteristics is superimposed on each channel. Even if it is done, the frequency characteristic of the head-related transfer function is not affected, and the feeling of localization does not change.
  • the array speaker apparatus 2002 does not superimpose the rear reverberation sound on each sound beam, but superimposes the rear reverberation sound on the sound that makes the virtual sound source perceived. Therefore, since the array speaker apparatus 2002 does not superimpose the rear reverberation sound having substantially the same frequency component on each sound beam regardless of the arrival direction, the audio signals of the sound beams may be similar and the sound images may be combined. Absent. As a result, the array speaker device 2002 prevents the localization of each sound beam from becoming unclear. In addition, since the sound beam perceives localization based on sound pressure from the direction of arrival, even if the frequency characteristics change due to superimposition of early reflected sound with different frequency characteristics for each direction of arrival, the feeling of localization does not change. .
  • the array speaker device 2002 provides the sound of the content by the initial reflected sound and the rear reverberant sound without impairing the sound localization effect that makes each sound beam and virtual sound source perceived. Can do.
  • the array speaker apparatus 2002 includes a set of the gain adjustment unit 2221 and the gain adjustment unit 2441, so that the level ratio between the initial reflected sound and the rear reverberation sound can be changed to a desired ratio of the listener.
  • the array speaker apparatus 2002 outputs a sound beam and a sound that makes a virtual sound source perceive for an audio signal of multi-channel surround sound, and gives a sound field effect. Therefore, the array speaker apparatus 2002 can give the sound of the content to the sound of the content while giving a sense of localization so as to surround the listener.
  • the rear reverberation sound generated by the rear reverberation sound generation unit 2442 is output from the woofers 2033L and 2033R after being superimposed on the sound causing the virtual sound source to be perceived, but the sound causing the virtual sound source to be perceived. It does not have to be superimposed on.
  • the audio signal of the rear reverberant sound generated by the rear reverberant sound generating unit 2442 may be input to the woofers 2033L and 2033R via the level adjusting units 2034L and 2034R without passing through the localization adding unit 2042.
  • FIG. 32 is a diagram for explaining the speaker set 2002A.
  • FIG. 33 is a part of a block diagram of the speaker set 2002A and the subwoofer 2003.
  • arrows indicate sound paths having directivity in the automobile interior 900.
  • the speaker set 2002A is different from the array speaker apparatus 2002 in that the directional speaker unit 2021 (2021Q, 2021R, 2021S, 2021T, 2021U) outputs sound having directivity. A description of the same configuration as that of the array speaker apparatus 2002 is omitted.
  • Each directional speaker unit 2021 is arranged according to a channel. That is, the directional speaker unit 2021S corresponding to the C channel is arranged in front of the listener.
  • the directional speaker unit 2021Q corresponding to the FL channel is arranged in front of the listener and on the left side.
  • the directional speaker unit 2021R corresponding to the FR channel is disposed in front of and right of the listener.
  • a directional speaker unit 2021T corresponding to the SL channel is disposed behind and to the left of the listener.
  • the directional speaker unit 2021U corresponding to the SR channel is disposed behind and to the right of the listener.
  • Each audio signal output from the level adjustment unit 2018 is input to a delay processing unit 2023 (2023FL, 2023FR, 2023C, 2023SR, 2023SL) as shown in FIG.
  • the delay processing unit 2023 performs delay processing according to the path length from each directional speaker unit 2021 to the listener so that the phase of sound having directivity is aligned near the listener.
  • Each audio signal output from the delay processing unit 2023 is input to each directional speaker unit 2021. Even with such a configuration, the speaker set 2002A can superimpose the initial reflected sound on the sound having directivity corresponding to each channel and deliver it to the listener.
  • the delay times of the delay processing unit 2060 and the delay processing unit 2023 are set so that sound having directivity and sound that perceives a virtual sound source are not output at the same timing.
  • FIG. 34 is a diagram for explaining an AV system 3001 provided with an array speaker device 3002.
  • FIG. 35 is a part of a block diagram of the array speaker device 3002 and the subwoofer 3003.
  • 36A is a block diagram of the localization adding unit 3042
  • FIG. 36B is a block diagram of the correction unit 3051.
  • FIG. 37 and FIG. 38 are diagrams showing the path of the sound beam output from the array speaker device 3002 and the localization position of the sound source by the sound beam, respectively.
  • FIG. 39 is a diagram for explaining calculation of the delay amount of the audio signal by the directivity control unit 3020.
  • the AV system 3001 includes an array speaker device 3002, a subwoofer 3003, and a television 3004.
  • Array speaker device 3002 is connected to subwoofer 3003 and television 3004.
  • the array speaker device 3002 receives an audio signal corresponding to an image reproduced on the television 3004 and an audio signal from a content player (not shown).
  • the array speaker device 3002 outputs a sound beam based on the input audio signal and allows the listener to localize the virtual sound source.
  • the array speaker device 3002 includes, for example, a rectangular parallelepiped housing.
  • the housing of the array speaker device 3002 includes, for example, 16 speaker units 3021A to 3021P and woofers 3033L and 3033R on the surface facing the listener.
  • the number of speaker units is not limited to 16, and may be, for example, 8 or the like.
  • the speaker units 3021A to 3021P, the woofer 3033L, and the woofer 3033R correspond to “a plurality of speakers” of the present invention.
  • Speaker units 3021A to 3021P are arranged in a row.
  • the speaker units 3021A to 3021P are arranged in order from the left side when viewing the array speaker device 3002 from the listener.
  • Woofer 3033L is arranged further to the left from speaker unit 3021A.
  • the woofer 3033R is disposed on the right side of the speaker unit 3021P.
  • the array speaker apparatus 3002 includes a decoder 3010 and a directivity control unit 3020 as shown in FIG.
  • the decoder 3010 is connected to a DIR (; Digital audio I / F Receiver) 3011, ADC (; Analog to Digital Converter) 3012, and HDMI (registered trademark; High Definition Multimedia Interface) receiver 3013.
  • DIR Digital audio I / F Receiver
  • ADC Analog to Digital Converter
  • HDMI registered trademark; High Definition Multimedia Interface
  • the DIR11 receives a digital audio signal transmitted by an optical cable or a coaxial cable.
  • the ADC 3012 converts the input analog signal into a digital signal.
  • the HDMI receiver 3013 receives an HDMI signal that conforms to the HDMI standard.
  • the decoder 3010 supports various data formats such as AAC (registered trademark), Dolby Digital (registered trademark), DTS (registered trademark), MPEG-1 / 2, MPEG-2 multichannel, and MP3.
  • the decoder 3010 converts the digital audio signals output from the DIR 3011 and the ADC 3012 into multi-channel audio signals (FL channel, FR channel, C channel, SL channel, and SR channel digital audio signals.
  • the audio signals are simply referred to as audio signals).
  • a digital audio signal The decoder 3010 extracts audio data from the HDMI signal (a signal conforming to the HDMI standard) output from the HDMI receiver 3013, decodes the audio data, and outputs the audio signal.
  • the decoder 3010 can convert the audio signal to various channels such as a 7-channel audio signal as well as a 5-channel audio signal.
  • the array speaker device 3002 divides the band of the audio signal output from the decoder 3010, outputs a high frequency (for example, 200 Hz or more) to the speaker units 3021A to 3021P, and outputs a low frequency (for example, less than 200 Hz) to the woofers 3033L and 3033R.
  • a high frequency for example, 200 Hz or more
  • a low frequency for example, less than 200 Hz
  • an HPF 3014 (3014FL, 3014FR, 3014C, 3014SR, 3014SL) and an LPF 3015 (3015FL, 3015FR, 3015C, 3015SR, 3015SL) are provided.
  • the cut-off frequencies of the HPF 3014 and the LPF 3015 are set so as to match the lower limit (200 Hz) of the reproduction frequency of the speaker units 3021A to 3021P.
  • the audio signal of each channel output from the decoder 3010 is input to the HPF 3014 and the LPF 3015, respectively.
  • the HPF 3014 extracts and outputs a high frequency component (200 Hz or more) of the input audio signal.
  • the LPF 3015 extracts and outputs a low frequency component (less than 200 Hz) of the input audio signal.
  • Each audio signal output from the HPF 3014 is input to the level adjustment unit 3018 (3018FL, 3018FR, 3018C, 3018SR, 3018SL).
  • the level adjustment unit 3018 is provided for adjusting the level of the sound beam for each channel.
  • the level adjustment unit 3018 adjusts and outputs the level of each audio signal.
  • the directivity control unit 3020 receives each audio signal output from the level adjustment unit 3018.
  • the directivity control unit 3020 distributes the input audio signals of the respective channels by the number of the speaker units 3021A to 3021P, and delays them by a predetermined delay time.
  • the delayed audio signal of each channel is converted into an analog audio signal by a not-shown DAC (Digital to Analog Converter), and then input to the speaker units 3021A to 3021P.
  • the speaker units 3021A to 3021P emit sound based on the input audio signals of the respective channels.
  • each of the output from the speaker units 3021A to 3021P is performed.
  • the sound strengthens the phase in a direction corresponding to the difference in the delay amount.
  • the sound beam is formed as a parallel wave traveling in a predetermined direction from the speaker units 3021A to 3021P.
  • the directivity control unit 3020 can also perform delay control so that the phases of the sounds output from the speaker units 3021A to 3021P are aligned at predetermined positions. In this case, each sound output from the speaker units 3021A to 3021P becomes an audio beam having the focus at the predetermined position.
  • the array speaker device 3002 may include an equalizer for each channel before or after the directivity control unit 3020 to adjust the frequency characteristics of each audio signal.
  • the audio signal output from the LPF 3015 is input to the woofer 3033L, the fa 3033R, and the subwoofer unit 3072.
  • the array speaker device 3002 further divides an audio signal (less than 200 Hz) other than the sound beam band into a band for the woofers 3033L and 3033R (for example, 100 Hz or more) and a band for the subwoofer unit 3072 (for example, less than 100 Hz).
  • HPF3030 (3030L, 3030R) and LPF3031 (3031L, 3031R) are provided.
  • the cutoff frequencies of the HPF 3030 and the LPF 3031 are set so as to match the upper limit (100 Hz) of the reproduction frequency of the subwoofer unit 3072.
  • the audio signal (less than 200 Hz) output from the LPF 3015 (3015FL, 3015C, 3015SL) is added by the adder 3016.
  • the audio signal added by the adder 3016 is input to the HPF 3030L and the LPF 3031L.
  • the HPF 3030L extracts and outputs a high frequency component (100 Hz or more) of the input audio signal.
  • the LPF 3031L extracts and outputs a low frequency component (less than 100 Hz) of the input audio signal.
  • the audio signal output from the HPF 3030L is input to the woofer 3033L via the level adjustment unit 3034L, the addition unit 3032L, and a DAC (not shown).
  • the audio signal output from the LPF 3031L is input to the subwoofer unit 3072 of the subwoofer 3003 via the level adjustment unit 3070F, the addition unit 3071, and the DAC (not shown).
  • the level adjustment unit 3034L and the level adjustment unit 3070F adjust the level of the input audio signal in order to adjust the level ratio of the sound beam, the sound output from the woofer 3033L, and the sound output from the subwoofer unit 3072. Adjust and output.
  • the audio signal output from the LPF 3015 (3015FR, 3015C, 3015SR) is added by the adder 3017.
  • the audio signal added by the adder 3017 is input to the HPF 3030R and the LPF 3031R.
  • the HPF 3030R extracts and outputs a high frequency component (100 Hz or more) of the input audio signal.
  • the LPF 3031R extracts and outputs a low frequency component (less than 100 Hz) of the input audio signal.
  • the audio signal output from the HPF 3030R is input to the woofer 3033R via the level adjustment unit 3034R, the addition unit 3032R, and a DAC (not shown).
  • the audio signal output from the LPF 3031R is input to the subwoofer unit 3072 via the level adjustment unit 3070G, the addition unit 3071, and the DAC (not shown).
  • the level adjuster 3034R and the level adjuster 3070G adjust the level of the input audio signal in order to adjust the level ratio of the sound beam, the sound output from the woofer 3033R, and the sound output from the subwoofer unit 3072. Adjust and output.
  • array speaker apparatus 3002 outputs sound beams (less than 200 Hz) other than the sound beam band from woofers 3033L and 3033R and subwoofer unit 3072 while outputting sound beams for each channel from speaker units 3021A to 3021P. .
  • the array speaker device 3002 includes a localization adding unit 3042, a crosstalk cancellation processing unit 3050, and delay processing units 3060L and 3060R.
  • the array speaker device 3002 divides the band of the audio signal output from the decoder 3010, outputs a high frequency (for example, 100 Hz or more) to the woofers 3033L and 3033R, and outputs a low frequency (for example, less than 100 Hz) to the subwoofer unit 3072.
  • HPF 3040 (3040FL, 3040FR, 3040C, 3040SR, 3040SL) and LPF 3041 (3041FL, 3041FR, 3041C, 3041SR, 3041SL) are provided.
  • the cutoff frequencies of the HPF 3040 and the LPF 3041 are set so as to match the upper limit (100 Hz) of the reproduction frequency of the subwoofer unit 3072, respectively.
  • the audio signal of each channel output from the decoder 3010 is input to the HPF 3040 and the LPF 3041, respectively.
  • the HPF 3040 extracts and outputs a high frequency component (100 Hz or more) of the input audio signal.
  • the LPF 3041 extracts and outputs a low frequency component (less than 100 Hz) of the input audio signal.
  • the array speaker device 3002 includes level adjusting units 3070A to 3070E in order to adjust the level ratio between the sound output from the woofers 3033L and 3033R and the sound output from the subwoofer unit 3072.
  • the level of each audio signal output from the LPF 3041 is adjusted by the level adjusting units 3070A to 3070E.
  • the audio signals whose levels have been adjusted by the level adjusters 3070A to 3070E are added by the adder 3071, respectively.
  • the audio signal added by the adder 3071 is input to the subwoofer unit 3072 via a DAC (not shown).
  • the array speaker device 3002 includes a level adjustment unit 3043 (3043FL, 3043FR, 3043C, 3043SR, 3043SL) in order to adjust the level of sound that makes a virtual sound source perceived for each channel.
  • a level adjustment unit 3043 (3043FL, 3043FR, 3043C, 3043SR, 3043SL) in order to adjust the level of sound that makes a virtual sound source perceived for each channel.
  • Each audio signal output from the HPF 3040 is input to the level adjustment unit 3043.
  • the level adjustment unit 3043 adjusts and outputs the level of each input audio signal.
  • Each audio signal output from the level adjustment unit 3043 is input to the localization adding unit 3042.
  • the localization adding unit 3042 performs processing for localizing each input audio signal to a virtual sound source position.
  • a head-related transfer function hereinafter referred to as HRTF
  • HRTF head-related transfer function
  • HRTF is an impulse response that expresses the loudness, arrival time, frequency characteristics, etc. from the virtual speaker installed at a certain position to the left and right ears.
  • the localization adding unit 3042 includes filters 3421L to 3425L and filters 3421R to 3425R for convolving the HRTF impulse response for each channel.
  • the FL channel audio signal (the audio signal output from the HPF 3040FL) is input to the filters 3421L and 3421R.
  • the filter 3421L adds an HRTF of the path from the position of the virtual sound source VSFL (see FIG. 37) on the left front of the listener to the left ear to the FL channel audio signal.
  • the filter 3421R adds an HRTF of the path from the position of the virtual sound source VSFL to the right ear to the FL channel audio signal.
  • the filter 3422L adds the HRTF of the path from the position of the virtual sound source VSFR in front of the listener to the left ear of the listener to the audio signal of the FR channel.
  • the filter 3422R adds the HRTF of the path from the position of the virtual sound source VSFR to the right ear to the audio signal of the FR channel.
  • Filters 3423L to 3425L add HRTFs in the path from the positions of the virtual sound sources VSC, VSSL, and VSSR corresponding to the C, SL, and SR channels to the listener's left ear to the audio signals of the C, SL, and SR channels.
  • the filters 3423R to 3425R add the HRTF of the path from the position of the virtual sound source VSC, VSSL, VSSR corresponding to the C, SL, SR channel to the right ear of the listener to the audio signal of the C, SL, SR channel.
  • the adding unit 3426L synthesizes the audio signals output from the filters 3421L to 3425L, and outputs them to the crosstalk cancellation processing unit 3050 as the audio signal VL.
  • the adder 3426R combines the audio signals output from the filters 3421R to 3425R, and outputs the synthesized audio signal VR to the crosstalk cancellation processing unit 3050.
  • the crosstalk cancellation processing unit 3050 emits sound from the woofer 3033L, emits the opposite phase component of crosstalk reaching the right ear from the woofer 3033R, and cancels the sound pressure at the position of the right ear, thereby canceling the sound of the woofer 3033L. Is suppressed from being heard in the right ear. Conversely, the crosstalk cancellation processing unit 3050 emits sound from the woofer 3033R, emits a reverse phase component of crosstalk reaching the left ear from the woofer 3033L, and cancels the sound pressure at the position of the left ear, thereby canceling the woofer. The sound of 3033R is suppressed from being heard by the left ear.
  • the crosstalk cancellation processing unit 3050 performs processing using the correction unit 3051 and the combining units 3052L and 3052R.
  • the correction unit 3051 includes direct correction units 3511L and 3511R and cross correction units 3512L and 3512R as shown in FIG.
  • the audio signal VL is input to the direct correction unit 3511L and the cross correction unit 3512L.
  • the audio signal VR is input to the direct correction unit 3511R and the cross correction unit 3512R.
  • the direct correction unit 3511L performs processing to make the listener perceive that the sound emitted from the woofer 3033L is emitted near the left ear.
  • the direct correction unit 3511L is set with a filter coefficient so that the sound output from the woofer 3033L can be heard flat at the position of the left ear.
  • the direct correction unit 3511L corrects the input audio signal VL and outputs an audio signal VLD.
  • the cross correction unit 3512R in combination with the synthesizing unit 3052L, outputs from the woofer 3033L a sound having the opposite phase of the sound that wraps around the left ear from the woofer 3033R, and cancels the sound pressure at the position of the left ear, thereby canceling the sound of the woofer 3033R. Is suppressed from being heard in the left ear.
  • the cross correction unit 3512R performs a process of making the listener perceive that the sound emitted from the woofer 3033L is emitted near the left ear.
  • the cross correction unit 3512R is set with a filter coefficient that prevents the sound output from the woofer 3033R from being heard at the position of the left ear.
  • the cross correction unit 3512R corrects the input audio signal VR and outputs an audio signal VRC.
  • the synthesizing unit 3052L makes the audio signal VRC out of phase and synthesizes it with the audio signal VLD.
  • the direct correction unit 3511R performs processing to make the listener perceive that the sound emitted from the woofer 3033R is emitted near the right ear.
  • filter coefficients are set such that the sound output from the woofer 3033R can be heard flat at the position of the right ear.
  • the direct correction unit 3511R corrects the input audio signal VR and outputs an audio signal VRD.
  • the cross correction unit 3512L in combination with the synthesizing unit 3052R, outputs from the woofer 3033R a sound having a phase opposite to that from the woofer 3033L and cancels the sound pressure at the position of the right ear, thereby canceling the sound of the woofer 3033L. Is suppressed from being heard in the right ear.
  • the cross correction unit 3512L performs processing to make the listener perceive that the sound emitted from the woofer 3033R is emitted near the right ear.
  • the cross correction unit 3512L is set with a filter coefficient so that the sound output from the woofer 3033L cannot be heard at the position of the right ear.
  • the cross correction unit 3512L corrects the input audio signal VL and outputs an audio signal VLC.
  • the synthesizing unit 3052R synthesizes the audio signal VLC with the audio signal VLC in reverse phase.
  • the audio signal output from the synthesis unit 3052L is input to the delay processing unit 3060L.
  • the audio signal is delayed by a predetermined time by the delay processing unit 3060L and input to the level adjusting unit 61L.
  • the audio signal output from the synthesis unit 3052R is input to the delay processing unit 3060R.
  • the delay processing unit 3060R delays the audio signal by the same delay time as the delay processing unit 3060L.
  • the delay time by the delay processing units 3060L and 3060R is set to be longer than the longest delay time among the delay times given to the audio signal to form the sound beam. Details of the delay time will be described later.
  • Level adjustment units 3061L and 3061R are provided to collectively adjust the sound levels that cause the virtual sound sources of all channels to be perceived.
  • the level adjustment units 3061L and 3061R adjust the level of each audio signal delayed by the delay processing units 3060L and 3060R.
  • the audio signals whose levels are adjusted by the level adjusting units 3061L and 3061R are input to the woofers 3033L and 3033R via the adding units 3032L and 3032R.
  • the audio signals outside the audio beam band (less than 200 Hz) output from the speaker units 3021A to 3021P are input to the adders 3032L and 3032R, the sound outside the audio beam band and the sound that localizes the virtual sound source Are output by the woofers 3033L and 3033R.
  • the array speaker device 3002 localizes the audio signal of each channel at a virtual sound source position.
  • an arrow indicates a path of an audio beam output from the array speaker device 3002.
  • the asterisk indicates the position of the sound source generated by the sound beam and the position of the virtual sound source.
  • the array speaker apparatus 3002 outputs five sound beams according to the number of channels of the input audio signal, as shown in FIG.
  • the C channel audio signal is delay-controlled so that, for example, the focal position is set on the wall in front of the listener. Then, the listener perceives that the sound source SC of the C channel audio signal is on the wall in front of the listener.
  • the audio signals of the FL and FR channels are delay-controlled so that the sound beam is focused on the left front wall of the listener and the right front wall of the listener.
  • Each sound beam based on the audio signals of the FL and FR channels is reflected once by the wall of the room R and reaches the position of the listener. Then, the listener perceives that the sound sources SFL and SFR of the audio signals of the FL and FR channels are on the left front wall and the right front wall of the listener.
  • the audio signals of the SL and SR channels are subjected to delay control, for example, so that the sound beam is directed toward the left side wall and the right side wall of the listener.
  • Each sound beam based on the audio signals of the SL and SR channels is reflected by the wall of the room R and reaches the left rear wall of the listener and the right rear wall of the listener.
  • Each sound beam is reflected again by the left rear wall of the listener and the right rear wall of the listener to reach the position of the listener. Then, the listener perceives that the sound sources VSSL, VSSSR of the audio signals of the SL and SR channels are on the left rear wall of the listener and the right rear wall of the listener.
  • the filters 3421L to 3425L and filters 3421R to 3425R of the localization adding unit 3042 are set so that the positions of the virtual speakers are substantially the same as the positions of the sound sources SFL, SFR, SC, SSL, and SSR, respectively. Then, as shown in FIG. 37, the listener perceives the virtual sound sources VSC, VSFL, VSFR, VSSL, VSSR at substantially the same positions as the positions of the sound sources SFL, SFR, SC, SSL, SSR.
  • the sound beam may diffuse when reflected depending on the wall.
  • the array speaker device 3002 can supplement the sense of localization due to the sound beam with the virtual sound source. Therefore, the array speaker device 3002 can improve the sense of localization as compared with the case where only the sound beam is used or the case where only the virtual sound source is used.
  • the sound sources SSL and SSR of the audio signals of the SL and SR channels are generated by reflecting each sound beam twice on the wall. Therefore, the SL and SR channel sound sources are less perceptible than the FL, C, and FR channel sound sources.
  • the array speaker apparatus 3002 can supplement the sense of localization of the SL and SR channels by the sound beam with the virtual sound sources VSSL and VSSR generated by the sound that directly reaches the listener's ear, and the sense of localization of the SL and SR channels can be compensated. There is no loss.
  • the array speaker device 3002 perceives a virtual sound source with sound that directly reaches the listener's ear, for example, even when the sound absorption of the wall of the room R is high and the sound beam is difficult to reflect. Therefore, a sense of orientation can be given to the listener.
  • the array speaker device 3002 lowers the gain of the level adjustment units 3061L and 3061R or increases the gain of the level adjustment unit 3018 to make the sound level perceived by the virtual sound source.
  • the level of the sound beam is increased compared to
  • the array speaker device 3002 increases the gain of the level adjustment units 3061L and 3061R or lowers the gain of the level adjustment unit 3018 to reduce the sound that makes the virtual sound source perceived in an environment where the sound beam is difficult to reflect. Reduce the sound beam level compared to the level.
  • the array speaker apparatus 3002 can adjust the ratio between the level of the sound beam and the level of the sound causing the virtual sound source to be perceived according to the environment.
  • the array speaker device 3002 does not change only the level of either the sound beam or the sound that causes the virtual sound source to be perceived, but simultaneously changes the level of both the sound beam and the sound that causes the virtual sound source to be perceived. You may go.
  • the array speaker device 3002 includes the level adjusting unit 3018 that adjusts the level of the sound beam for each channel, and the level adjusting unit 3043 that adjusts the level of the sound that causes the virtual sound source to be perceived for each channel.
  • the array speaker device 3002 includes a set of a level adjustment unit 3018 and a level adjustment unit 3043 for each channel, thereby changing, for example, the ratio between the level of the sound beam and the level of the sound that causes the virtual sound source to be perceived only for the FL channel. be able to.
  • the array speaker apparatus 3002 can give a sense of localization by increasing the sound perceived by the virtual sound source VSFL, for example, even in an environment where the sound source SFL is not easily localized by the sound beam.
  • the delay processing units 3060L and 3060R delay the sound that causes the virtual sound source to be perceived so that the sound that causes the virtual sound source to perceive does not hinder the formation of the sound beam.
  • time DT The time for which the delay processing units 3060L and 3060R delay the audio signal (hereinafter, time DT) is calculated based on the time for which the directivity control unit 3020 delays the audio signal.
  • the delay time DT may be calculated by other functional units.
  • the delay time DT is calculated as follows. In the example shown in FIG. 39, the sound beam for generating the sound source SFR will be described.
  • the directivity control unit 3020 calculates a distance DP from the speaker unit 3021P to the focal point F of the sound beam.
  • the distance DP is calculated by a trigonometric function. That is, it is calculated
  • DP Sqrt ((XF-XP) 2 + (YF-YP) 2 + (ZF-ZP) 2 )
  • Sqrt is a function that takes a square root
  • coordinates (XF, YF, ZF) indicate the position of the focal point F.
  • the coordinates (XP, YP, ZP) indicate the position of the speaker unit 3021P, and are set in the array speaker device 3002 in advance.
  • the coordinates (XF, YF, ZF) are set, for example, via a user interface provided in the array speaker device 3002.
  • the directivity control unit 3020 calculates the distance DP, and obtains a difference distance DDP from the reference distance Dref by the following equation.
  • the reference distance Dref is a distance from the reference position S to the focal point F of the array speaker device 3002.
  • the coordinates of the reference position S are set in the array speaker device 3002 in advance.
  • the directivity control unit 3020 calculates the difference distances DDA to DDO for the remaining speaker units 3021A to 3021O. That is, directivity control unit 3020 calculates difference distances DDA to DDP for all speaker units 3021A to 3021P.
  • the directivity control unit 3020 selects the longest differential distance DDMAX and the shortest differential distance DDMIN from the differential distances DDA to DDP.
  • the delay time T corresponding to the distance difference DDDIF between the difference distance DDMAX and the difference distance DDMIN is calculated by dividing the distance difference DDDIF by the speed of sound.
  • the delay time T is calculated for the sound beam that generates the sound source SFR.
  • the sound beam having the largest output angle is formed from the sound output most slowly among all the sound beams.
  • the output angle of the sound beam is an angle ⁇ formed by the X axis and a line connecting from the reference position S to the focal point F. Therefore, the directivity control unit 3020 identifies the sound beam having the largest output angle, and obtains the delay time T (hereinafter referred to as delay time TMAX) corresponding to the sound beam.
  • the directivity control unit 3020 sets the delay time DT longer than the delay time TMAX and gives the delay time to the delay processing units 3060L and 3060R.
  • the sound that causes the virtual sound source to be perceived is output later than the sound that forms each sound beam. That is, the woofers 3033L and 3033R do not output sound as part of the speaker array including the speaker units 3021A to 3021P.
  • the sound that perceives the virtual sound source is less likely to hinder the formation of the sound beam.
  • the array speaker device 3002 can improve the sense of localization without impairing the sense of localization of the sound source by the sound beam.
  • delay processing units 3060L and 3060R may be provided before the localization adding unit 3042, or between the localization adding unit 3042 and the crosstalk cancellation processing unit 3050.
  • the directivity control unit 3020 may be configured to give the number of samples to be delayed instead of giving the delay time DT to the delay processing units 3060L and 3060R.
  • the number of samples to be delayed is calculated by multiplying the delay time DT by the sampling frequency.
  • FIG. 40 (A) is a diagram showing an array speaker device 3002A according to Modification 1 of the array speaker device 3002 according to the present embodiment.
  • FIG. 40B is a diagram showing an array speaker device 3002B according to Modification 2 of the array speaker device 3002. A description of the same configuration as that of the array speaker device 3002 is omitted.
  • Array speaker device 3002A is different from array speaker device 3002 in that sounds output from woofer 3033L and woofer 3033R are output from speaker unit 3021A and speaker unit 3021P, respectively.
  • array speaker apparatus 3002A outputs sound that makes a virtual sound source perceived and sound outside the band of the sound beam (100 Hz or more and less than 200 Hz) from speaker units 3021A and 3021P at both ends of speaker units 3021A to 3021P. It is.
  • the speaker unit 3021A and the speaker unit 3021P are arranged farthest among the speaker units 3021A to 3021P. Therefore, array speaker apparatus 3002A can perceive a virtual sound source.
  • the array speaker device 3002 may not include the speaker units 3021A to 3021P and the woofers 3033L and 3033R in one housing.
  • each speaker unit may be provided in an individual casing, and the casings may be arranged in a line.
  • the input multi-channel audio signals are respectively delayed and distributed to a plurality of speakers, and the input multi-channel audio signals are filtered based on the head-related transfer function. Any aspect that inputs to a plurality of speakers belongs to the technical scope of the present invention.
  • FIG. 41 is a block diagram showing a configuration of an array speaker device 3002C according to a modification.
  • the components common to the array speaker device 3002 are denoted by the same reference numerals, and the description thereof is omitted.
  • the array speaker device 3002C is different from the array speaker device 3002 in that instead of the delay processing unit 3060L and the delay processing unit 3060R, a delay processing unit 3062A to a delay processing unit 3062P are provided after the directivity control unit 3020.
  • Delay processing unit 3062A to delay processing unit 3062P delay audio signals supplied to speaker units 3021A to 3021P, respectively. That is, in the delay processing units 3062A to 3062P, the audio signals input from the directivity control unit 3020 to the speaker units 3021A to 3021P and the audio signals input from the localization adding unit 3042 to the woofer 3033L and the woofer 3033R are displayed. The audio signal is delayed so that it is delayed.
  • the delay processing unit 3060L and the delay processing unit 3060R delay the sound that perceives the virtual sound source so that the sound that perceives the virtual sound source does not hinder the formation of the sound beam.
  • the device 3002C is a mode in which the delay processing units 3062A to 3062P delay the sound forming the sound beam so that the sound forming the sound beam does not hinder the sound causing the virtual sound source to be perceived. For example, when the distance between the listening position and the wall is long, the environment is made of wall material with low acoustic reflectivity, or the number of speakers is small, the sound beam reflection from the wall is weak, and the localization feeling due to the sound beam is weak There is.
  • the sound forming the sound beam may interfere with the sound that causes the virtual sound source to be perceived. Therefore, in the array speaker device 3002C, the sound that forms the sound beam is delayed so as not to disturb the sound that causes the virtual sound source to be perceived, and is reproduced with a delay from the sound that causes the virtual sound source to be perceived.
  • the delay processing unit 3062A to the delay processing unit 3062P are provided in the subsequent stage of the directivity control unit 3020, but the delay for delaying the audio signal of each channel in the previous stage of the directivity control unit 3020. It is good also as an aspect which provides a process part.
  • the array speaker device includes a delay processing unit 3060L and a delay processing unit 3060R, and a delay processing unit 3062A to a delay processing unit 3062P can be considered.
  • the intensity of reflection from the wall surface can be measured using a microphone installed at the listening position, for example, by turning an audio beam of a test sound such as white noise.
  • a test sound such as white noise.
  • the array speaker device can measure the intensity of the reflection of the sound beam from the wall by detecting the level of the sound beam of the collected test sound.
  • the array speaker device determines that the sound beam is strongly reflected from the wall when the level of the collected sound beam is equal to or higher than a predetermined threshold, and delays the sound that causes the virtual sound source to be perceived.
  • the array speaker device determines that the reflection of the sound beam from the wall is weak when the level of the collected sound beam is less than a predetermined threshold, and delays the sound forming the sound beam.
  • the speaker device of the present invention delays and distributes an input unit to which a plurality of channels of audio signals are input, a plurality of speakers, and a plurality of channels of audio signals input to the input unit to the plurality of speakers.
  • a directivity control unit that outputs a plurality of sound beams to the plurality of speakers, and a filtering process based on a head-related transfer function is performed on any of the plurality of channels of audio signals input to the input unit,
  • a localization adding unit that inputs to a speaker, a first level adjusting unit that adjusts the levels of the audio signal of each channel of the localization adding unit and the audio signal of each channel of the sound beam, and the first level adjusting unit
  • setting means for setting the level.
  • the speaker device of the present invention is configured such that the sense of localization by the sound beam is supplemented by the virtual sound source. Thereby, a feeling of localization can be improved as compared with the case where only the sound beam is used or the case where only the virtual sound source is used.
  • the speaker device of the present invention detects a level difference at which the sound beam of each channel reaches the listening position, and adjusts the level of each channel of the localization adding unit and each channel of the sound beam based on the detected level difference. To do.
  • the level of the localization adding unit is set higher than the other channels, and the effect of localization addition by the virtual sound source is strengthened. Also, in the speaker device of the present invention, since a localization feeling due to an audio beam is present even for a channel in which the localization effect by the virtual sound source is set strongly, there is a possibility that only a specific channel becomes a virtual sound source and a sense of incongruity occurs. And the connection between channels is maintained.
  • the speaker device of the present invention further includes a microphone installed at the listening position and a detecting unit that detects a level at which the sound beam of each channel reaches the listening position, and the detecting unit includes the directivity control.
  • a test signal is input to the unit and a test sound beam is output to the plurality of speakers, a level at which the test sound beam is input to the microphone is measured, and the setting unit is based on a measurement result of the detection unit.
  • the level ratio in the first level adjustment unit is set.
  • the level of each channel of the localization adding unit and each channel of the sound beam is automatically adjusted together with the output angle of the sound beam of each channel by simply installing a microphone at the listening position and performing the measurement.
  • the speaker device of the present invention further includes a comparison unit that compares levels of audio signals of a plurality of channels input to the input unit, and the setting unit is configured to compare the level adjustment unit based on a comparison result of the comparison unit. Set the level at.
  • the channel to which a high-level signal is input sets the localization adding unit at a higher level than the other channels, and the localization effect by the virtual sound source is strengthened so that the sound image is clearly localized.
  • the comparison unit compares the levels of the front channel audio signal and the surround channel audio signal, and the setting unit sets the level in the first level adjustment unit based on the comparison result of the comparison unit. To do.
  • the surround channel needs to reach the sound beam from behind the listening position, and needs to reflect the sound beam twice on the wall. For this reason, the surround channel may not have a clear sense of localization compared to the front channel. Therefore, for example, when the level of the surround channel becomes relatively high, the level of the localization addition unit is set high, and the localization effect of the virtual sound source is strengthened, so that the localization feeling of the surround channel is maintained. When the level of the front channel is relatively high, the localization feeling by the sound beam is set strongly. On the other hand, if the level ratio of the localization addition part is lowered when the level of the surround channel is relatively low, the surround channel may become difficult to hear, so the level of the surround channel is relatively low. In this case, the level ratio of the localization adding section may be increased, and the level ratio of the localization adding section may be decreased when the level of the surround channel becomes relatively high.
  • the comparing means may divide a plurality of channels of audio signals input to the input unit into predetermined bands, and compare the signal levels of the divided bands.
  • the speaker device of the present invention may include a volume setting receiving unit that receives volume settings of the plurality of speakers, and the setting unit may set the level in the level adjusting unit based on the volume setting.
  • the volume setting (master volume setting) of multiple speakers is lowered, the level of sound reflected from the walls will be reduced, resulting in a sound with no thickness, and the connection between channels may be lost, resulting in a reduction in surround sound. There is. Therefore, it is preferable to maintain the connection between channels and maintain the surround feeling by setting the level of the localization adding unit higher as the master volume setting becomes lower and strengthening the localization addition effect by the virtual sound source.
  • the speaker device of the present invention delays and distributes the audio signals of a plurality of channels input to the input unit, the plurality of speakers, and the input unit to which the audio signals of a plurality of channels are input to the plurality of speakers.
  • a directivity control unit that outputs sound beams to the plurality of speakers, and a filtering process based on a head-related transfer function is performed on each of the plurality of channels of audio signals input to the input unit and input to the plurality of speakers.
  • a localization addition unit is performed.
  • the localization adding unit of the speaker device sets the direction of the virtual sound source based on the head-related transfer function in the direction between the arrival directions of the plurality of sound beams as viewed from the listening position. That is, the direction of the virtual sound source based on the head-related transfer function is set in the direction between a plurality of beams like a phantom sound source.
  • the speaker device of the present invention clearly locates the sound source in the intended direction using the virtual sound source based on the head-related transfer function that does not depend on the listening environment such as the acoustic reflectivity of the wall while using the sense of localization by the sound beam. be able to.
  • the direction of the virtual sound source by the head-related transfer function is set to the same direction as the phantom sound source generated by a plurality of beams, for example.
  • the localization of the phantom sound source due to the sound beam can be compensated and the sound source can be localized more clearly.
  • the localization adding unit sets a direction of a virtual sound source based on the head-related transfer function in a direction that is symmetrical with respect to the listening position as a central axis with respect to at least one arrival direction of the sound beam. It is good also as an aspect.
  • the sound source is localized in a symmetrical direction when viewed from the listening position.
  • the speaker device includes a microphone installed at a listening position, a test signal input to the directivity control unit, and a test audio beam output to the plurality of speakers, the test audio beam being transmitted to the microphone.
  • the apparatus may further comprise detection means for measuring an input level and beam angle setting means for setting an output angle of the sound beam based on a peak of the level measured by the detection means.
  • the localization adding unit sets the direction of the virtual sound source based on the head-related transfer function based on the level peak measured by the detection means. Thereby, the direction of the virtual sound source is automatically set together with the output angle of the sound beam of each channel simply by installing the microphone at the listening position and performing the measurement.
  • the speaker device of the present invention includes an input unit to which an audio signal is input, a first sound emission unit that emits sound based on the input audio signal, and a second sound emission that emits sound based on the input audio signal.
  • a localization adding unit that performs filtering on the audio signal input to the input unit based on a head-related transfer function and inputs the filtered signal to the first sound emitting unit, and an initial reflected sound characteristic of the input audio signal.
  • the localization adding unit receives the audio signal output from the rear reverberation adding unit, and the directivity control unit receives the audio signal output from the initial reflected sound adding unit.
  • the rear reverberation adding unit does not add the characteristic of the initial reflected sound to the sound perceived by the virtual sound source, but adds the characteristic of the initial reflected sound only to the sound output from the second sound emitting unit. Therefore, the speaker device prevents a change in the frequency characteristics of the sound that causes the virtual sound source to perceive due to the addition of the characteristics of the initial reflected sound having different frequency characteristics for each direction of arrival. Thereby, the frequency characteristic of the head-related transfer function is maintained for the sound that causes the virtual sound source to be perceived.
  • the speaker device of the present invention does not impair the sense of localization due to the sound that causes the virtual sound source to be perceived even when the sound field effect is provided by the initial reflection sound and the rear reverberation sound.
  • the speaker device may include a level adjusting unit that adjusts the levels of the initial reflected sound of the initial reflected sound adding unit and the rear reverberant sound of the rear reverberant sound adding unit.
  • the level of the initial reflection sound and the level of the rear reverberation sound can be set to a desired ratio of the listener.
  • the audio signal may be a multi-channel surround sound audio signal.
  • the speaker device can provide the sound field effect while virtually positioning the audio signal so as to surround the listener.
  • the first sound emitting unit may output a sound having directivity.
  • the speaker device may output an audio beam as a sound having directivity with the following configuration.
  • the first sound emitting unit includes a stereo speaker to which the audio signal of the localization adding unit is input, and the second sound emitting unit delays the speaker array and the audio signal input to the input unit, respectively.
  • a directivity control unit that distributes to the speaker array may be used.
  • a sound beam is output as follows as sound having directivity.
  • a speaker array including a plurality of speaker units emits sound based on the audio signal delayed and distributed by the directivity control unit.
  • the directivity control unit delay-controls the audio signal so that the phases of sounds output from the plurality of speaker units are aligned at predetermined positions. As a result, the sound output from each of the plurality of speaker units is strengthened at a predetermined position and becomes a sound beam having directivity.
  • the localization adding unit performs a filtering process so that the virtual sound source is localized at a position where the listener perceives the sound source with sound having directivity or a position near the position.
  • the speaker device improves the feeling of localization compared to the case where only sound having directivity is used or the case where only a virtual sound source is used.
  • the rear reverberation sound adding unit does not add the characteristic of the rear reverberation sound to the sound having directivity, but adds the characteristic of the rear reverberation sound only to the sound perceiving the virtual sound source emitted from the first sound emission unit. Therefore, since the speaker device does not add the characteristic of the rear reverberant sound to the sound having directivity, the localization of the sound having directivity is prevented from being unclear by being pulled to the center part of the reverberation.
  • the speaker device of the present invention includes an input unit to which an audio signal is input, a plurality of speakers, a directivity control unit that delays and distributes the audio signals input to the input unit to the plurality of speakers, A localization adding unit that performs filtering on the audio signal input to the input unit based on a head-related transfer function and inputs the filtered signal to the plurality of speakers.
  • the plurality of speakers emit sound based on the audio signal delayed and distributed by the directivity control unit.
  • the directivity control unit delay-controls the audio signal so that the phases of sounds output from the plurality of speaker units are aligned at predetermined positions. As a result, the sound output from each of the plurality of speaker units is strengthened at a predetermined position and becomes a sound beam having directivity. The listener perceives the sound source by listening to the sound beam.
  • the localization adding unit performs a filtering process so that the virtual sound source is localized at a position where the listener perceives the sound source by the sound beam or a position near the position.
  • the speaker device of the present invention can improve the sense of localization as compared with the case where only the sound beam is used or the case where only the virtual sound source is used.
  • the speaker device of the present invention can improve the feeling of localization by adding a feeling of localization by a virtual sound source without impairing the feeling of localization of the sound source by the sound beam.
  • the speaker device of the present invention includes a delay processing unit that outputs the audio signal with a delay before or after the localization adding unit or the directivity control unit.
  • the sound that should form the sound beam may be out of phase by the sound that perceives the virtual sound source. That is, if the sound that causes the virtual sound source to be perceived is output at the same time as the sound that should form the sound beam, the formation of the sound beam may be hindered by the sound that causes the virtual sound source to be perceived. Therefore, the speaker device of the present invention outputs a sound that causes the virtual sound source to be perceived later than the sound that forms the sound beam. As a result, the sound that perceives the virtual sound source is less likely to hinder the formation of the sound beam.
  • the delay processing unit is provided before or after the localization adding unit, and delays and outputs the audio signal by a delay amount larger than a maximum delay amount delayed by the directivity control unit. It is preferable.
  • the delay processing unit is provided before or after the directivity control unit, and the audio signal input to the plurality of speakers from the directivity control unit is compared with the audio signal input to the plurality of speakers from the localization adding unit. It is preferable that the audio signal be delayed and output so as to be delayed. Accordingly, the sound that forms the sound beam is delayed so as not to hinder the sound that causes the virtual sound source to be perceived, and the sound that is delayed than the sound that causes the virtual sound source to be perceived can be reproduced.
  • the speaker device may include a level adjusting unit that adjusts the level of the audio signal of the directivity control unit and the audio signal of the localization adding unit.
  • ⁇ ⁇ Virtual sound sources are perceived as sounds that reach the listener directly, and are therefore less dependent on the environment.
  • the sound beam depends on the environment because it uses reflection on the wall, but can give a sense of localization more than a virtual sound source.
  • the audio signal may be a multi-channel surround sound audio signal.
  • the sound source is perceived by the listener using reflection on the wall, but the sound image may be blurred due to reflection.
  • the rear channel audio signal is less likely to be localized than the front channel because the audio beam uses two wall reflections.
  • the speaker device since the speaker device also allows a virtual sound source to be perceived by sound that directly reaches the listener, it can give a sense of localization in the rear channel as much as the front channel.
  • the plurality of speakers includes a speaker array to which the audio signal of the directivity control unit is input and a stereo speaker to which the audio signal of the localization adding unit is input, and the audio signal input to the input unit
  • a band dividing unit that divides the frequency band into a high frequency region and a low frequency region, and outputs each of them, and the directivity control unit receives the high frequency audio signal output from the band frequency dividing unit, and the stereo speaker May be a mode in which a low-frequency audio signal output from the band dividing unit is input.
  • the stereo speaker is used both for the output of sound that makes a virtual sound source perceive and the output of sound that is lower than the band of the sound beam.
  • the low frequency range where it is difficult to form an audio beam is compensated by the stereo speaker.
  • the audio signal processing device includes an input step in which audio signals of a plurality of channels are input, and the audio signals of the plurality of channels input in the input step are respectively delayed and distributed to a plurality of speakers.
  • a directivity control step for outputting a plurality of sound beams to a plurality of speakers, and performing a filtering process based on a head-related transfer function on at least one of the plurality of channels of audio signals input in the input step.
  • a localization adding step to be input to.
  • a first level adjusting step for adjusting the level of the audio signal of each channel filtered in the localization adding step and the audio signal of each channel of the sound beam, and the first level adjusting step And a setting step for setting a level at.
  • a detection step of detecting a level at which the sound beam of each channel reaches the listening position by a microphone installed at the listening position is further provided.
  • the detecting step from the plurality of speakers based on the input test signal
  • An audio signal processing method for measuring the level at which the output test sound beam is input to the microphone, and setting the level in the first level adjustment step based on the measurement result in the detection step in the setting step It is.
  • a comparison step for comparing levels of audio signals of a plurality of channels input in the input step is further provided, and the level in the level adjustment step is set in the setting step based on a comparison result in the comparison step.
  • This is an audio signal processing method.
  • the comparison step compares the levels of the front channel audio signal and the surround channel audio signal
  • the setting step determines the level in the first level adjustment step based on the comparison result in the comparison step. Is an audio signal processing method for setting.
  • the comparison step is an audio signal processing method that divides a plurality of channels of audio signals input in the input step into predetermined bands and compares the signal levels of the divided bands.
  • the audio signal processing method further includes a volume setting reception step of receiving volume settings of the plurality of speakers, and the setting step sets the level in the first level adjustment step based on the volume setting.
  • the localization adding step is an audio signal processing method for setting the direction of the virtual sound source based on the head-related transfer function between the arrival directions of a plurality of sound beams as viewed from the listening position.
  • the head further includes a phantom processing step of outputting audio signals of the same channel to a plurality of sound beams to localize the phantom sound source, and in the localization adding step, the head in a direction corresponding to the localization direction of the phantom sound source
  • a phantom processing step of outputting audio signals of the same channel to a plurality of sound beams to localize the phantom sound source
  • the head in a direction corresponding to the localization direction of the phantom sound source
  • an initial reflected sound adding step for adding an initial reflected sound characteristic to the input audio signal
  • a rear reverberant sound adding step for adding a rear reverberant sound characteristic to the input audio signal
  • the localization is further provided.
  • An adding step processes the audio signal processed in the rear reverberation adding step
  • the directivity control step processes the audio signal processed in the initial reflected sound adding step. It is.
  • the audio signal processing further includes a second level adjustment step of adjusting a level between the initial reflection sound processed in the initial reflection sound addition step and the rear reverberation sound processed in the rear reverberation sound addition step. Is the method.
  • a part of the plurality of speakers is a stereo speaker to which the audio signal processed in the localization adding step is input, and the other part of the plurality of speakers is processed in the directivity control step.
  • the audio signal processing method is a speaker array to which an audio signal is input.
  • the audio signal processing method further includes a delay processing step of delaying and outputting the audio signal before or after the processing in the localization adding step or the directivity control step.
  • the delay processing step is provided before or after the processing of the localization adding step,
  • the audio signal is output by delaying the audio signal by a delay amount larger than a maximum delay amount delayed in the directivity control step.
  • the delay processing step is provided before or after the processing of the directivity control step.
  • the delay processing step the multi-channel audio that is processed by the directivity control step and input to the plurality of speakers.
  • the audio signal is delayed and output so that the signal is processed by the localization adding step and delayed from the audio signals input to the plurality of speakers.
  • the method further includes a band dividing step of dividing the band of the audio signal input in the input step into a high frequency band and a low frequency band, and outputting each of the bands, and the plurality of speakers are processed by the directivity control step.
  • the directivity control step the high frequency band processed in the band dividing step is processed.
  • the audio signal processing method an audio signal is processed, and the low frequency audio signal processed in the band dividing step is input to the stereo speaker.
  • voice signal which can localize a sound source clearly using the localization by a virtual sound source, giving a feeling of localization with both a sound beam and a virtual sound source, utilizing the characteristic of a sound beam.
  • a processing method can be provided.
  • array speaker device 1002A ... array speaker device, 1002A ... speaker device, 1003 ... subwoofer, 1004 ... TV , 1007 ... microphone, 1010 ... decoder, 1011 ... input unit, 1 014, 1015 ... Filter processing unit, 1020 ... Beam processing unit, 1032 ... Addition processing unit, 1033L, 1033R ... Woofer, 1035 ... Control unit, 1036 ... User I / F, 1040 ... Virtual processing unit, 2001 ... AV system, 2002 , 2002A ... array speaker device, 2003 ... subwoofer, 2004 ... television, 2010 ... decoder, 2011 ... DIR, 2012 ... ADC, 2013 ...
  • HDMI receiver 2014FL, 2014FR, 2014C, 2014SR, 2014SL ... HPF, 2015FL, 2015FR, 2015C, 2015SR, 2015SL ... LPF, 2016, 2017 ... adder, 2018 ... level adjuster, 2020 ... directivity controller, 2021A to 2021P ... speaker unit, 2021Q, 2021 R, 2021S, 2021U, 2021T ... Directional speaker unit, 2022 ... Initial reflection sound processing unit, 2221 ... Gain adjustment unit, 2222 ... Initial reflection sound generation unit, 2223 ... Synthesis unit, 2030L, 2030R ... HPF, 2031L, 2031R ... LPF, 2032L, 2032R ... adder, 2033L, 2033R ...
  • HPF HPF, 3031L, 3031R ... LPF, 3032L, 3032R ... adder, 3033L, 3033R ... woofer

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  • Physics & Mathematics (AREA)
  • Engineering & Computer Science (AREA)
  • Acoustics & Sound (AREA)
  • Signal Processing (AREA)
  • Stereophonic System (AREA)
  • Circuit For Audible Band Transducer (AREA)

Abstract

La présente invention concerne un dispositif de haut-parleur pouvant localiser clairement une source sonore au moyen de la localisation d'une source sonore virtuelle tout en tirant le meilleur parti de la caractéristique d'un faisceau sonore. Un dispositif de haut-parleur (2) est équipé des éléments suivants : une unité d'entrée (11) au niveau de laquelle sont appliqués des signaux audio d'une pluralité de canaux ; une pluralité de haut-parleurs (21A-21P, 33L, 33R) ; une unité de contrôle de directivité (20) qui amène la pluralité de haut-parleurs à émettre une pluralité de faisceaux sonores en retardant les signaux audio de la pluralité de canaux, qui ont été appliqués au niveau de l'unité d'entrée (11), et en répartissant les signaux audio vers la pluralité de haut-parleurs ; et une unité d'addition de localisation (42) qui effectue un traitement de filtrage sur la base d'une fonction de transfert associé à la tête sur au moins l'un des signaux audio de la pluralité de canaux, qui ont été appliqués au niveau de l'unité d'entrée (11), et qui applique les signaux audio au niveau de la pluralité des haut-parleurs.
PCT/JP2014/071686 2013-08-19 2014-08-19 Dispositif de haut-parleur et procédé de traitement de signal audio WO2015025858A1 (fr)

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CN201480002397.6A CN104641659B (zh) 2013-08-19 2014-08-19 扬声器设备和音频信号处理方法
EP14838464.7A EP3038385B1 (fr) 2013-08-19 2014-08-19 Dispositif de haut-parleur et procédé de traitement de signal audio
US14/428,227 US9674609B2 (en) 2013-08-19 2014-08-19 Speaker device and audio signal processing method
US15/472,591 US10038963B2 (en) 2013-08-19 2017-03-29 Speaker device and audio signal processing method

Applications Claiming Priority (10)

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JP2013-169755 2013-08-19
JP2013169755 2013-08-19
JP2013-269162 2013-12-26
JP2013-269163 2013-12-26
JP2013269162A JP6405628B2 (ja) 2013-12-26 2013-12-26 スピーカ装置
JP2013269163A JP6287191B2 (ja) 2013-12-26 2013-12-26 スピーカ装置
JP2013272352A JP6287202B2 (ja) 2013-08-19 2013-12-27 スピーカ装置
JP2013272528A JP6287203B2 (ja) 2013-12-27 2013-12-27 スピーカ装置
JP2013-272528 2013-12-27
JP2013-272352 2013-12-27

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Cited By (4)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CN110083328A (zh) * 2019-03-29 2019-08-02 北京奇艺世纪科技有限公司 一种音频播放方法、装置、电子设备及存储介质
CN111246343A (zh) * 2019-10-04 2020-06-05 友达光电股份有限公司 扬声系统、显示装置以及音场重建方法
CN113286251A (zh) * 2020-02-19 2021-08-20 雅马哈株式会社 音信号处理方法及音信号处理装置
WO2023130206A1 (fr) * 2022-01-04 2023-07-13 Harman International Industries, Incorporated Système de haut-parleurs multicanal et procédé associé

Citations (7)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
JP2006340330A (ja) * 2005-06-06 2006-12-14 Yamaha Corp スピーカアレイ装置及びスピーカアレイ装置の音声ビーム設定方法
JP2007110744A (ja) * 2006-11-27 2007-04-26 Yamaha Corp スピーカアレイ装置
JP2007266967A (ja) * 2006-03-28 2007-10-11 Yamaha Corp 音像定位装置およびマルチチャンネルオーディオ再生装置
JP2009017137A (ja) * 2007-07-03 2009-01-22 Yamaha Corp スピーカアレイ装置
JP2009531926A (ja) * 2006-03-31 2009-09-03 コーニンクレッカ フィリップス エレクトロニクス エヌ ヴィ データ処理装置及び方法
JP2009218888A (ja) * 2008-03-11 2009-09-24 Yamaha Corp 放音システム
JP2012235456A (ja) * 2012-04-20 2012-11-29 Pioneer Electronic Corp 音声信号処理装置及び音声信号処理プログラム

Patent Citations (7)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
JP2006340330A (ja) * 2005-06-06 2006-12-14 Yamaha Corp スピーカアレイ装置及びスピーカアレイ装置の音声ビーム設定方法
JP2007266967A (ja) * 2006-03-28 2007-10-11 Yamaha Corp 音像定位装置およびマルチチャンネルオーディオ再生装置
JP2009531926A (ja) * 2006-03-31 2009-09-03 コーニンクレッカ フィリップス エレクトロニクス エヌ ヴィ データ処理装置及び方法
JP2007110744A (ja) * 2006-11-27 2007-04-26 Yamaha Corp スピーカアレイ装置
JP2009017137A (ja) * 2007-07-03 2009-01-22 Yamaha Corp スピーカアレイ装置
JP2009218888A (ja) * 2008-03-11 2009-09-24 Yamaha Corp 放音システム
JP2012235456A (ja) * 2012-04-20 2012-11-29 Pioneer Electronic Corp 音声信号処理装置及び音声信号処理プログラム

Cited By (5)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CN110083328A (zh) * 2019-03-29 2019-08-02 北京奇艺世纪科技有限公司 一种音频播放方法、装置、电子设备及存储介质
CN111246343A (zh) * 2019-10-04 2020-06-05 友达光电股份有限公司 扬声系统、显示装置以及音场重建方法
CN113286251A (zh) * 2020-02-19 2021-08-20 雅马哈株式会社 音信号处理方法及音信号处理装置
CN113286251B (zh) * 2020-02-19 2023-02-28 雅马哈株式会社 音信号处理方法及音信号处理装置
WO2023130206A1 (fr) * 2022-01-04 2023-07-13 Harman International Industries, Incorporated Système de haut-parleurs multicanal et procédé associé

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