JP6287191B2 - Speaker device - Google Patents

Description

  The present invention relates to a speaker device that outputs a sound beam having directivity.

  2. Description of the Related Art Conventionally, there is known 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 1 and 2).

  The array speaker device of Patent Document 1 uses a C channel sound beam and a sound beam reflected on a wall to reach the listener, and outputs the same signal at a predetermined ratio to localize the phantom sound source. Is. 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.

  In addition, the array speaker device of Patent Document 2 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.

JP 2005-159518 A JP 2010-213031 A

  However, depending on the listening environment, there are cases where reflection from the wall is not good (for example, an environment made of a wall material having low acoustic reflectivity). In addition, the sound volume or the frequency characteristic of the beam reflected from the wall surface for each channel does not completely match each sound beam. Therefore, with a phantom sound source using an audio beam, it is difficult to clearly localize the sound source in the intended direction.

  Therefore, an object of the present invention is to provide a speaker device capable of clearly localizing a sound source in an intended direction even when an audio beam is used.

  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. And 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.

  And the localization addition part of a speaker apparatus sets the direction of the virtual sound source based on the said head-related transfer function between the arrival directions of a several sound beam seeing from a 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.

  As a result, 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.

  Note that 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. As a result, 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. In this case, the sound source is localized in a symmetrical direction when viewed from the listening position.

  The speaker device according to the present invention 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. In this case, 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 can clearly localize a sound source in an intended direction even when an audio beam is used.

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. It 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. It is the figure which showed the relationship between an audio beam and channel setting. It is a block diagram which shows the structure of a virtual processing part. It is a block diagram which shows the structure of a localization addition part and a correction | amendment part. It is a figure explaining the sound field which the array speaker apparatus 2 produces | generates. It is a figure explaining the sound field which the array speaker apparatus 2 produces | generates. It is a block diagram which shows the structure of the array speaker apparatus in the case of using a phantom sound source together. FIG. 11A is a block diagram illustrating a configuration of the phantom processing unit. FIG. 11B is a diagram showing a correspondence table between designated angles and gain ratios. FIG. 11C is a diagram showing a correspondence table between the designated angle and the head-related transfer function. It is a block diagram which shows the structure of a phantom process part. It is a figure explaining the sound field which array speaker device 2A generates. It is a figure explaining the sound field which array speaker device 2A generates. It is a figure which shows 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).

  As shown in FIG. 1, 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 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. In this example, the speaker units 21A to 21P, the woofer 33L, and the woofer 33R correspond to “a plurality of speakers” of the present invention.

  Note that the number of speaker units is not limited to 16, and may be, for example, 8 or the like. Also, 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 to 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 a 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, a control unit 35, and a user. An I / F 36 is provided.

  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 an 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.

  The HPF 14FL, HPF 14FR, HPF 14C, HPF 14SL, and HPF 14SR extract and output the high frequencies of the input audio signals of the respective channels. 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 a low frequency (for example, less than 200 Hz) of an 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).

  The output signals of the 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 a high frequency 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 via the user I / F 36.

  The output signals of the LPF 15FR, LPF 15C, and LPF 15SR 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 via the user I / F 36.

  The audio signal output from the HPF 30L is input to the woofer 33L via the addition processing unit 32. Similarly, 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 31 </ b> L and the audio signal output from the LPF 31 </ b> R are added to a monaural signal by the addition processing unit 70 and input to the subwoofer 3. Although not shown, 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.

  On the other hand, 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.

  The HPF 40FL, HPF 40FR, HPF 40C, HPF 40SL, and HPF 40SR extract and output the high frequencies of the input audio signals of the respective channels. 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 via the user I / F 36. 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. Further, 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.

  The LPF 41FL, LPF 41FR, LPF 41C, LPF 41SL, and LPF 41SR extract and output the low frequencies of the input audio signals of the respective channels. 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. In 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.

  Next, the beamization processing unit 20 will be described. 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 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 the audio signals of the respective channels to speaker units 21A to 21P. Audio signals for the distributed speaker units 21A to 21P are combined by the combining unit 92 and supplied to the speaker units 21A to 21P. At this time, 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 are output as sound beams 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.

  In addition, 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 configured so that the phase of each sound output from the speaker units 21A to 21P is 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 2 or reflected on the wall of the room. For example, as shown in FIG. 5C, 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. Also, 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. In addition, 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. As shown in FIG. 5A, when the listener places the microphone 7 at the listening position and operates the user I / F 36 (or a remote controller (not shown)) to instruct the setting of the sound beam, the control unit 35 An audio beam composed of a test signal (for example, white noise) is output to the beam processing unit 20.

  The control unit 35 passes from the left direction parallel to the front surface of the array speaker device 2 (referred to as a -90 degree direction) to the array speaker device 2 through a direction perpendicular to the front surface of the array speaker device 2 (referred to as a 0 degree direction). The sound beam is turned to the right direction (referred to as 90 degree direction) parallel to the front surface of No. 2. 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 θ of the sound beam, and the sound beam is picked up by the microphone 7 at a predetermined angle. .

  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. And the control part 35 allocates the output angle of each channel and audio | voice beam of a multichannel audio signal based on the peak component of the level of an 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 θ3a at the highest level is assigned as the output angle of the sound beam of the C channel. In addition, the 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 θ2a close to the C channel and close to the −90 degree direction is assigned as an SL channel sound beam, and an angle θ4a 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 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. 5B, the angle θ1a closest to the −90 degrees direction is assigned as the FL channel sound beam, and the angle θ5a closest to the 90 degrees direction is assigned as the output angle of the FR channel sound beam. In this way, the control unit 35 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 level peak measured by the detection unit. Implement setting means.

  As described above, as shown in FIG. 5C, setting is performed such that the sound beam reaches from the surroundings at the position of the listener (microphone 7).

  Next, the virtual processing unit 40 will be described. 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 adjustment unit 43FL, the gain adjustment unit 43FR, the gain adjustment unit 43C, the gain adjustment unit 43SL, and the gain adjustment unit 43SR adjust the gain of the audio signal of each channel. The gain of each gain adjustment unit is set by the control unit 35 based on the detection result of the test sound beam, for example. 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. In the other gain adjustment unit, a gain corresponding to the level difference from the C channel is set. For example, when the gain 0.1 of the gain adjustment unit 43C is set with the C channel detection level G1 = 1.0, and the FR channel detection level G3 = 0.6, the gain of the gain adjustment unit 43FR is set to 0.4. If 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. In the examples of FIGS. 5A, 5B, and 5C, the control unit 35 turns the sound beam of the test signal so that the sound beam of each channel reaches the listening position. Although an example of detection is shown, the listener uses the user I / F 36 to manually instruct the control unit 35 to output an audio beam, and the gain adjustment unit 43FL, the gain adjustment unit 43FR, the gain adjustment unit 43C, The levels of the gain adjustment unit 43SL and the gain adjustment unit 43SR may be set manually. 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. In order to localize as a virtual sound source, a head-related transfer function (hereinafter referred to as HRTF) indicating a transfer function between a predetermined position and a listener's ear is used.

  The HRTF is an impulse response that expresses the volume, arrival time, frequency characteristics, and the like of the sound 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 illustrating a 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.

  For example, an 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. Similarly, for each 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 a process of making the listener perceive that the sound output from the woofer 33L is emitted near the left ear. In the direct correction unit 511L, 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. In the direct correction unit 511R, 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 providing 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 imparting frequency characteristics of 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 synthesis 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. Alternatively, 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. In 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. Note that the addition processing unit 32 may include a configuration of a gain adjustment unit that changes the addition ratio of these audio signals. Similarly, the audio signal output from the delay processing unit 60R is input to the woofer 33R via the addition processing unit 32. In 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.

  Next, the sound field generated by the array speaker device 2 will be described with reference to FIG. In FIG. 8A, the solid line arrow indicates the path of the sound beam output from the array speaker device 2. In FIG. 8A, a white star indicates the position of the sound source generated by the sound beam, and a black star indicates the position of the virtual sound source.

  In the example of FIG. 8A, the array speaker device 2 outputs five sound beams. The C-channel audio signal is set to an audio beam that is focused at a position behind the array speaker device 2. Thus, the listener perceives that the sound source SC is in front of the listener.

  Similarly, in the audio signal of the FL channel, 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.

  However, in the example of FIG. 8A, the distance between the right front wall and the listening position is longer than the distance between the left front wall and the listening position. Therefore, the sound source SFR is perceived closer to the rear than the sound source SFL. Therefore, the localization adding unit 42 is set between the C channel sound beam and the FR channel sound beam. In this example, the localization adding unit 42 sets the direction of the virtual sound source VSFR to be symmetrical 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 set manually by the listener using the user I / F 36, but can be automatically set as follows, for example.

  As shown in FIG. 8B, the control unit 35 determines the symmetry of the peaks existing in the regions on both sides across the peak angle θa3 set for the C channel.

  For example, if the allowable error is ± 10 degrees and the control section 35 is −10 degrees ≦ θa2 + θa4 ≦ 10 degrees, the arrival directions of the sound beams of the SL channel and the SR channel are determined to be symmetrical. Similarly, the control unit 35 determines that the arrival directions of the audio beams of the FL channel and the FR channel are symmetrical if −10 degrees ≦ θa1 + θa5 ≦ 10 degrees.

  FIG. 8B shows an example in which the value of θa1 + θa5 exceeds the allowable error. Therefore, the control unit 35 instructs the localization adding unit 42 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).

  In the example of FIG. 8B, the direction of the virtual sound source VSFR is set to θa5 ′ that is symmetrical to θa1 across the central axis (here, θa3 = 0 degree). In the other channels, 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.

  As a result, the array speaker device 2 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 reflectance of the wall, while using the sense of localization by the sound beam. be able to. Further, in the examples of FIGS. 8A and 8B, since the sound source is localized in a bilaterally symmetric direction when viewed from the listening position, a more ideal listening mode is obtained.

  Next, FIG. 9A is a diagram illustrating an example in the case where the SR channel reaches further forward than the SL channel. In this case, 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. Similarly to the above, the control unit 35 determines whether or not −10 degrees ≦ θa2 + θa4 ≦ 10 degrees, for example, with an allowable error of ± 10 degrees. FIG. 9B shows an example in which the value of θa2 + θa4 exceeds the allowable error. Therefore, the control unit 35 instructs the localization adding unit 42 to set the direction of the virtual sound source between the arrival directions of the two sound beams.

  Also in this case, 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. To do. In the example of FIG. 9B, the direction of the virtual sound source VSSL is set to θa2 ′ that is symmetrical to θa4 across the central axis (here, θa3). In other channels, 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.

  As a result, the sound source is localized in a bilaterally symmetric direction as viewed from the listening position for the surround channel, which is a more ideal listening mode.

  In particular, since the sound source SSL and the sound source SSR are generated by reflecting the sound beam twice on the wall, there may be a case where a clear localization feeling cannot be obtained as compared with the channel on the front side. However, 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 sound source can be clearly heard in a more ideal direction. Can be localized.

  Next, FIG. 10 is a block diagram showing a configuration of the array speaker device 2A when the phantom sound source is used together. The components common to the array speaker device 2 in FIG. 2 are denoted by the same reference numerals and description thereof is omitted.

  The array speaker device 2A differs from the array speaker device 2 in that a phantom processing unit 90 is provided. The phantom processing unit 90 distributes the audio signal of each channel among the audio signals input from the filter processing unit 14 to its own channel and other channels, thereby localizing a specific channel (generating a phantom sound source). )

  FIG. 11A is a block diagram illustrating a configuration of the phantom processing unit 90. FIG. 11B is a diagram showing a correspondence table between designated angles and gain ratios. FIG. 11C is a diagram showing a correspondence table between designated angles and filter coefficients (head-related transfer functions provided by the localization adding unit 42). The phantom processing unit 90 includes a gain adjusting unit 95FL, a gain adjusting unit 96FL, a gain adjusting unit 95FR, a gain adjusting unit 96FR, a gain adjusting unit 95SL, a gain adjusting unit 96SL, a gain adjusting unit 95SR, a gain adjusting unit 96SR, an adding unit 900, An adder 901 and an adder 902 are provided.

  An audio signal of the FL channel is input to the gain adjustment unit 95FL and the gain adjustment unit 96FL. The FR channel audio signal is input to the gain adjustment unit 95FR and the gain adjustment unit 96FR. The SL channel audio signal is input to the gain adjustment unit 95SL and the gain adjustment unit 96SL. The SR channel audio signal is input to gain adjustment unit 95SR and gain adjustment unit 96SR.

  The gain ratio of the FL channel audio signal is adjusted by the gain adjusting unit 95FL and the gain adjusting unit 96FL, and then input to the adding unit 901 and the adding unit 900, respectively. The gain ratio of the audio signal of the FR channel is adjusted by the gain adjusting unit 95FR and the gain adjusting unit 96FR and input to the adding unit 902 and the adding unit 900, respectively. The gain ratio of the SL channel audio signal is adjusted by the gain adjusting unit 95SL and the gain adjusting unit 96SL, and the signals are input to the beam forming processing unit 20 and the adding unit 901, respectively. The gain ratio of the SR channel audio signal is adjusted by the gain adjustment unit 95SR and the gain adjustment unit 96SR, and is input to the beam forming processing unit 20 and the addition unit 902, respectively.

  The gain of each gain adjustment unit is set by the control unit 35. As shown in FIG. 11B, the control unit 35 reads the association table stored in the memory (not shown), and reads the gain ratio associated with the designated angle. In this example, the control unit 35 controls the gain ratio between the sound beam of the FR channel arriving from the front right of the listening position and the sound beam of the C channel arriving from the front of the listening position. Control the direction of the phantom sound source.

  An example in which a phantom sound source and a virtual sound source are used together will be described with reference to FIG. In this example, when the arrival direction θa5 of the sound beam of the FR channel is 80 degrees (80 degrees to the right when viewed from the listening position), the designated angle is in the direction of 40 degrees (40 degrees to the right when viewed from the listening position). A case where the FR channel phantom sound source is localized will be described.

  The control unit 35 has a designated angle of 40 degrees, an FR channel audio beam arrival direction θa5 (FR angle) of 80 degrees, and a C channel audio beam arrival direction θa3 (C angle) of 0 degrees. The gains of the gain adjustment unit 95FR and the gain adjustment unit 96FR corresponding to the gain ratio 100 * (40/80) = 50 are read out. In this case, the control unit 35 sets the gain of the gain adjustment unit 95FR to 0.5 and the gain of the gain adjustment unit 96FR to 0.5. As a result, as shown in FIG. 12, the phantom sound source can be localized in the direction of 40 degrees to the right between the FR channel audio beam and the C channel audio beam reaching from the front of the listening position. Here, the gain ratio is set so that the gain (0.5) of the gain adjusting unit 95FR + the gain (0.5) of the gain adjusting unit 96FR = 1.0 (so that the gain is constant). Although an example is shown, the gain may be set so that the power is constant. In this example, the gain of the gain adjustment unit 95FR and the gain of the gain adjustment unit 96FR are −3 dB (about 0.707).

  Then, the control unit 35 reads out from the table shown in FIG. 11C the filter coefficient at which the virtual sound source is localized in the direction of the designated angle of 40 degrees, and sets it in the localization adding unit 42. 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 36, but can also be automatically set using the measurement result of the test sound beam described above. For example, the arrival direction θa1 of the FL channel audio beam is −60 degrees (60 degrees to the left when viewed from the listening position), and the FR channel phantom sound source is localized in a direction symmetrical to the arrival direction of the FL channel audio beam. The specified angle is 60 degrees to the right. In this case, if the FR angle is 80 degrees and the C angle is 0 degrees, the gains of the gain adjustment unit 95FR and the gain adjustment unit 96FR corresponding to the gain ratio 100 * (60/80) = 75 are read. Therefore, the control unit 35 sets the gain of the gain adjustment unit 95FR to 0.75 and the gain of the gain adjustment unit 96FR to 0.25.

  In this way, the array speaker device 2A 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 localizes the phantom sound source. be able to.

  In particular, since the surround channel uses the sound beams (for example, FL channel sound beam and SL channel sound beam) to localize the phantom sound source, the localization is clearer than when the front channel is localized as the phantom sound source. A feeling may not be obtained. However, the array speaker device 2A 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 33L and the woofer 33R, and can localize the phantom sound source more clearly. Can do.

  The array speaker device 2A is suitable for a case where an audio signal of a larger number of channels is localized with a smaller number of sound beams. FIG. 13 is a diagram showing an example in which a 7.1-channel audio signal is localized using five sound beams. In addition to 5.1 channel surround (C, FL, FR, SL, SR, LFE), 7.1 channel surround includes 2 channels (SBL, SBR) to be played from behind the listener. In this example, the array speaker device 2A sets the SBL channel to an audio beam that focuses on the left rear wall position of the room R, and the audio beam that focuses the SBR channel on the right left rear wall position of the room R. Set to.

  Further, the array speaker apparatus 2A 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). Similarly, 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.

  Then, the array speaker device 2A 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.

  Thus, even when a large number of channels are localized with a small number of sound beams, the array speaker device 2A can supplement the sense of localization with a virtual sound source generated by sound that reaches the listener's ear directly by the woofer 33L and the woofer 33R. A large number of channels can be localized more clearly.

  Next, FIG. 14A is a diagram showing an array speaker device 2B according to a modification. A description of the same configuration as that of the array speaker device 2 is omitted.

  Array speaker device 2B differs 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 2B 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 21 </ b> A and the speaker unit 21 </ b> P are the speaker units arranged at the extreme ends 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 perceived.

  Further, 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 one housing. For example, as in a speaker set 2C shown in FIG. 14B, each speaker unit may be provided in an individual casing, and the casings may be arranged side by side.

DESCRIPTION OF SYMBOLS 1 ... AV system 2 ... Array speaker apparatus 2A ... Array speaker apparatus 2A ... Speaker apparatus 3 ... Subwoofer 4 ... Television 7 ... Microphone 10 ... Decoder 11 ... Input part 14, 15 ... Filter processing part 20 ... Beam processing part 32 ... Addition Processing unit 33L, 33R ... Woofer 35 ... Control unit 36 ... User I / F
40 ... Virtual processing section

Claims (4)

An input unit for inputting multi-channel audio signals;
Multiple speakers,
A directivity control unit that outputs audio beams to the plurality of speakers by delaying and distributing the plurality of channels of audio signals input to the input unit to the plurality of speakers;
A localization adding unit that performs filtering processing based on a head-related transfer function to audio signals of a plurality of channels input to the input unit and inputs the audio signals to the plurality of speakers;
A speaker device comprising:
The speaker apparatus, wherein the localization adding unit sets a direction of a virtual sound source based on the head-related transfer function between arrival directions of a plurality of sound beams as viewed from a listening position. A phantom processing unit that outputs audio signals of the same channel to a plurality of sound beams and localizes the phantom sound source;
The speaker device according to claim 1, wherein the localization adding unit sets the direction of the virtual sound source based on the head-related transfer function in a direction corresponding to the localization direction of the phantom sound source.   The localization adding unit sets the direction of the 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. The speaker device according to claim 1, wherein the speaker device is characterized. A microphone installed at the listening position;
Detecting means for inputting a test signal to the directivity control unit to output a test sound beam to the plurality of speakers, and measuring a level at which the test sound beam is input to the microphone;
Beam angle setting means for setting the output angle of the sound beam based on the level peak measured by the detection means,
The said localization addition part sets the direction of the virtual sound source based on the said head-related transfer function based on the peak of the level which the said detection means measured, The Claim 1 thru | or 3 characterized by the above-mentioned. Speaker device.

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