JP2015128208A - Speaker device - Google Patents

Abstract

PROBLEM TO BE SOLVED: To provide a speaker device that does not damage a sense of localization even when outputting sound allowing a listener to sense a virtual sound source and giving a sound field effect.SOLUTION: An initial reflected sound adding section adds characteristics of a reflected sound to an audio signal to be inputted in a second sound emitting section; and a rear reverberation sound adding section adds characteristics of a rear reverberation sound to an audio signal to be inputted in a first sound emitting section. Precisely, the reflected sound adding section does not add characteristics of an initial reflected sound to a sound allowing a listener to sense a virtual sound source, but adds the characteristics of the initial reflected sound only to a sound to be outputted from the second sound emitting section. Therefore, a speaker device can prevent frequency characteristics of the sound allowing the listener to sense the virtual sound source from being changed by addition of the characteristics of the initial reflected sound whose frequency characteristics differ depending on arrival direction. This allows the sound allowing the listener to sense the virtual sound source to maintain frequency characteristics of a head-elated transfer function.

Description

  The present invention relates to a speaker device that outputs a sound that allows a virtual sound source to be perceived to virtually localize an audio signal.

  2. Description of the Related Art Conventionally, there is known a speaker device that outputs a sound that perceives a virtual sound source in which an audio signal is virtually localized and a sound having directivity to localize the audio signal.

  For example, the array speaker device disclosed in Patent Document 1 outputs a plurality of directional sound beams (sound beams) by delaying an audio signal and distributing it to a plurality of speaker units. When the listener listens to the sound beam, the listener perceives that the sound source exists at the focus of the sound beam. When the sound beam is reflected, for example, on a wall and reaches the listener, the listener perceives that a sound source is present in the direction of the wall. As a result, the audio signal is localized in the direction of the wall.

  In addition, the array speaker device disclosed in Patent Document 1 changes the frequency characteristics of an audio signal in order to output a sound that causes a virtual sound source to be perceived for a channel in which an audio beam cannot reach the listener. More specifically, 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.

JP2008-227803

  The array speaker device shown in Patent Document 1 is limited to channels in which an audio beam cannot reach, and the audio signal is virtually localized, and the audio beam and the sound that makes the virtual sound source be perceived are exclusively output. In order to improve the feeling, it is conceivable to output both the sound beam and the sound that makes the virtual sound source perceived simultaneously.

  Conventionally, it has been proposed to add a sound field effect to the sound of content. 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.

  Here, 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 walls 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. Hereinafter, a sound simulating an actual early reflection sound is simply referred to as an initial reflection sound, and a sound simulating an actual rear reverberation sound is simply referred to as a rear reverberation sound.

  However, in a speaker device that outputs both sound having directivity and sound that causes a virtual sound source to be perceived on the same channel, when the initial reflected sound and the rear reverberation sound are superimposed on sound having a directivity and sound that causes the virtual sound source to be perceived The following problems arise:

  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. In addition, when 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.

  Therefore, an object of the present invention is to provide a speaker device that outputs sound that makes a virtual sound source perceived and does not impair the sense of localization even when a sound field effect is applied.

  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. An initial reflected sound adding unit for adding, and a rear reverberant adding unit for adding a characteristic of the reverberant sound to the input audio signal.

  The initial reflected sound adding unit adds the characteristics of the initial reflected sound to the audio signal input to the second sound emitting unit, and the rear reverberant adding unit is input to the first sound emitting unit. The rear reverberation characteristic is added to the audio signal.

  The rear reverberation adding unit does not add the characteristic of the initial reflected sound to the sound perceived by the virtual sound source, and 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.

  Thus, 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 reflected sound and the rear reverberant sound.

  The speaker device may further include a level adjusting unit that adjusts a level ratio between the initial reflected sound of the initial reflected sound adding unit and the rear reverberant sound of the rear reverberant sound adding unit.

  Thereby, 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.

  Thereby, the speaker device can provide the sound field effect while virtually locating the audio signal so as to surround the listener.

  Further, the first sound emitting unit may output a sound having directivity. For example, 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.

  In this aspect, an audio beam is output as a sound having directivity as follows. 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 by sound having directivity or a position near the position. As a result, 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, and adds the characteristic of the rear reverberation sound only to the sound that perceives 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.

  According to the present invention, even when a sound field effect is applied, the speaker device does not add the characteristics of the initial reflected sound having different frequency characteristics for each direction of arrival to the sound perceived by the virtual sound source. The frequency characteristic of the function is maintained and the sense of localization is not impaired.

It is a figure for demonstrating the AV system 1 provided with the array speaker apparatus 2 which concerns on this embodiment. It is a part of block diagram of the array speaker apparatus 2 and the subwoofer 3 which concern on this embodiment. 3 is a block diagram of an initial reflected sound processing unit 22 and a rear reverberation sound processing unit 44. FIG. It is a schematic diagram which shows the example of the impulse response measured in the concert hall. 4 is a block diagram of a localization adding unit 42 and a correction unit 51. FIG. It is a figure for demonstrating the sound which the array speaker apparatus 2 outputs. It is a figure for demonstrating the speaker set 2A which concerns on the modification of the array speaker apparatus 2 which concerns on this embodiment. It is a part of block diagram of the speaker set 2A and the subwoofer 3.

  An array speaker device 2 according to the present embodiment will be described with reference to FIGS. FIG. 1 is a diagram for explaining an AV system 1 including an array speaker device 2. FIG. 2 is a part of a block diagram of the array speaker device 2 and the subwoofer 3. FIG. 3A is a block diagram of the initial reflected sound processing unit 22, and FIG. 3B is a block diagram of the rear reverberation sound processing unit 44. FIG. 4 is a schematic diagram showing an example of the impulse response actually measured in the concert hall. FIG. 5A is a block diagram of the localization adding unit 42, and FIG. 5B is a block diagram of the correction unit 51. FIG. 6 is a diagram for explaining the sound output by the array speaker device 2.

  The AV system 1 includes an array speaker device 2, a subwoofer 3, and a television 4. 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 a video reproduced on the television 4 and a content audio signal from a content player (not shown). The array speaker device 2 outputs a sound beam having directivity and a sound that makes 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.

  First, the sound beam and the output of the initial reflected sound will be described. As shown in FIG. 1, the array speaker device 2 includes, for example, a rectangular parallelepiped housing. The housing of the array speaker device 2 includes, for example, 16 speaker units 21A to 21P and woofers 33L and 33R (corresponding to the first sound emission unit of the present invention) on the surface facing the listener. However, the number of speaker units is not limited to 16, and may be, for example, 8 or the like.

  The speaker units 21A to 21P are each arranged in a line. The speaker units 21 </ b> A to 21 </ b> P are arranged in order from the left side when viewing the array speaker device 2 from the listener. The woofer 33L is disposed further to the left from the speaker unit 21A. The woofer 33R is disposed on the right side of the speaker unit 21P.

  As shown in FIG. 2, the array speaker device 2 includes a decoder 10 and a directivity control unit 20. In addition, the group of the speaker units 21A to 21P and the directivity control unit 20 corresponds to the second sound emitting unit of the present invention.

  The decoder 10 is connected to a DIR (; Digital audio I / F Receiver) 11, an ADC (; Analog to Digital Converter) 12, and an HDMI (registered trademark; High Definition Multimedia Interface) receiver 13.

  The DIR 11 receives a digital audio signal transmitted through an optical cable or a coaxial cable. The ADC 12 converts the input analog signal into a digital signal. The HDMI receiver 13 receives an HDMI signal that conforms to the HDMI standard.

  The decoder 10 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 10 converts the digital audio signals output from the DIR 11 and the ADC 12 into multi-channel audio signals (FL channel, FR channel, C channel, SL channel, and SR channel digital audio signals. Hereinafter, the audio signals are simply referred to as audio signals). , And a digital audio signal). The decoder 10 extracts audio data from the HDMI signal (a signal conforming to the HDMI standard) output from the HDMI receiver 13, decodes it into an audio signal, and outputs the audio signal. However, the decoder 10 is not limited to the 5-channel audio signal, and can also convert the audio signal into various channels such as a 7-channel audio signal.

  The array speaker device 2 divides the band of the audio signal output from the decoder 10, outputs the high frequency (for example, 200 Hz or more) to the speaker units 21A to 21P, and outputs the low frequency (for example, less than 200 Hz) to the woofers 33L and 33R. In addition, HPF 14 (14FL, 14FR, 14C, 14SR, 14SL) and LPF 15 (15FL, 15FR, 15C, 15SR, 15SL) are provided for output to subwoofer unit 72. The cut-off frequencies of the HPF 14 and the LPF 15 are set so as to match the lower limit (200 Hz) of the reproduction frequency of the speaker units 21A to 21P.

  The audio signals of the respective channels output from the decoder 10 are input to the HPF 14 and the LPF 15 respectively. The HPF 14 extracts and outputs a high frequency component (200 Hz or more) of the input audio signal. The LPF 15 extracts and outputs a low frequency component (less than 200 Hz) of the input audio signal.

  As shown in FIG. 2, the array speaker device 2 includes an initial reflected sound processing unit 22 in order to add a sound field effect of the initial reflected sound to the sound of the content. Each audio signal output from the HPF 14 is input to the initial reflected sound processing unit 22. The initial reflected sound processing unit 22 superimposes the audio signal of the initial reflected sound on each input audio signal, and outputs it to the level adjusting unit 18 (18FL, 18FR, 18C, 18SR, 18SL).

  More specifically, the initial reflected sound processing unit 22 includes a gain adjusting unit 221, an initial reflected sound generating unit 222, and a synthesizing unit 223, as shown in FIG. Each audio signal input to the initial reflected sound processing unit 22 is input to the gain adjustment unit 221 and the synthesis unit 223. The gain adjustment unit 221 inputs the level of each input audio signal and the gain adjustment unit 441 (see FIG. 3B) in order to adjust the level ratio between the initial reflected sound and the rear reverberation 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 222.

  The initial reflected sound generation unit 222 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.

  As shown in FIG. 4, the actual initial reflected sound is generated until a predetermined time (for example, within 300 msec) elapses from the generation of the direct sound (point of time 0 in the schematic diagram of FIG. 4). 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. 4, for example. Then, the audio signal of the initial reflection sound generated by the initial reflection sound generation unit 222 is distributed and output to the audio signal of each channel according to the actual direction of arrival 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 14 to the synthesizing unit 223).

  Each audio signal output from the initial reflected sound generation unit 222 is input to the synthesis unit 223. The synthesizer 223 outputs an audio signal obtained by synthesizing the audio signal input from the HPF 14 and the audio signal input from the initial reflected sound generator 222 to the level adjuster 18 for each channel. As a result, the initial reflected sound is superimposed on the direct sound (corresponding to the audio signal directly input from the HPF 14 to the synthesis unit 223). 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 adjusting unit 18 is provided for adjusting the level of the sound beam for each channel. The level adjusting unit 18 adjusts and outputs the level of each audio signal.

  The directivity control unit 20 receives each audio signal output from the level adjustment unit 18. The directivity control unit 20 distributes the input audio signals of the respective channels by the number of the speaker units 21A to 21P, 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 21A to 21P. The speaker units 21 </ b> A to 21 </ b> P emit sound based on the input audio signal of each channel.

  When the directivity control unit 20 performs delay control so that a difference in delay amount given to an audio signal input to an adjacent speaker unit among the speaker units 21A to 21P is constant, each output from the speaker units 21A to 21P is performed. The sound strengthens the phase in a direction corresponding to the difference in the delay amount. As a result, the sound beam is formed as a parallel wave traveling in a predetermined direction from the speaker units 21A to 21P.

  The directivity control unit 20 can also perform delay control so that the phases of the sounds output from the speaker units 21A to 21P are aligned at predetermined positions. In this case, each sound output from the speaker units 21 </ b> A to 21 </ b> P becomes an audio beam having the predetermined position as a focal point.

  The array speaker device 2 may be provided with an equalizer for each channel before or after the directivity control unit 20 to adjust the frequency characteristics of each audio signal.

  The audio signal output from the LPF 15 is input to the woofer 33L or 33R and the subwoofer unit 72.

  The array speaker device 2 divides an audio signal (less than 200 Hz) other than the audio beam band into a band for the woofers 33L and 33R (for example, 100 Hz or more) and a band for the subwoofer unit 72 (for example, less than 100 Hz). HPF30 (30LL, 30R) and LPF31 (31L, 31R) are provided. The cut-off frequencies of the HPF 30 and the LPF 31 are set so as to match the upper limit (100 Hz) of the reproduction frequency of the subwoofer unit 72, respectively.

  Audio signals (less than 200 Hz) output from the LPF 15 (15FL, 15C, 15SL) are added by the adder 16. The audio signal added by the adding unit 16 is input to the HPF 30L and the LPF 31L. The HPF 30L extracts and outputs a high frequency component (100 Hz or more) of the input audio signal. The LPF 31L extracts and outputs a low frequency component (less than 100 Hz) of the input audio signal. The audio signal output from the HPF 30L is input to the woofer 33L via the level adjustment unit 34L, the addition unit 32L, and a DAC (not shown). The audio signal output from the LPF 31L is input to the subwoofer unit 72 of the subwoofer 3 via the level adjustment unit 70F, the addition unit 71, and the DAC (not illustrated). The level adjuster 34L and the level adjuster 70F 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 33L, and the sound output from the subwoofer unit 72. Adjust and output.

  The audio signal output from the LPF 15 (15FR, 15C, 15SR) is added by the adding unit 17. The audio signal added by the adding unit 17 is input to the HPF 30R and the LPF 31R. The HPF 30R extracts and outputs a high frequency component (100 Hz or more) of the input audio signal. The LPF 31R extracts and outputs a low frequency component (less than 100 Hz) of the input audio signal. The audio signal output from the HPF 30R is input to the woofer 33R via the level adjustment unit 34R, the addition unit 32R, and a DAC (not shown). The audio signal output from the LPF 31R is input to the subwoofer unit 72 via the level adjustment unit 70G, the addition unit 71, and a DAC (not shown). The level adjustment unit 34R and the level adjustment unit 70G 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 33R, and the sound output from the subwoofer unit 72. Adjust and output.

  As described above, the array speaker device 2 outputs the sound beam on which the initial reflected sound is superimposed from the speaker units 21A to 21P for each channel, and outputs sound (less than 200 Hz) other than the sound beam band to the woofers 33L and 33R. And output from the subwoofer unit 72.

  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. In addition, in a stage preceding the reverberation sound processing unit 44, as a mode including only the HPF 40FL, HPF 40FR, HPF 40C, HPF 40SL, and HPF 40SR, the low range may not be output to the subwoofer 3.

  Next, localization of the virtual sound source and output of the rear reverberation sound will be described. As shown in FIG. 2, the array speaker device 2 includes a rear reverberation processing unit 44, a localization adding unit 42, a crosstalk cancellation processing unit 50, and delay processing units 60L and 60R.

  The array speaker device 2 divides the band of the audio signal output from the decoder 10, outputs the high frequency (for example, 100 Hz or more) to the woofers 33 </ b> L and 33 </ b> R, and outputs the low frequency (for example, less than 100 Hz) to the subwoofer unit 72. In order to do so, HPF 40 (40FL, 40FR, 40C, 40SR, 40SL) and LPF 41 (41FL, 41FR, 41C, 41SR, 41SL) are provided. The cutoff frequencies of the HPF 40 and the LPF 41 are set so as to match the upper limit (100 Hz) of the reproduction frequency of the subwoofer unit 72, respectively.

  The audio signals of each channel output from the decoder 10 are input to the HPF 40 and the LPF 41, respectively. The HPF 40 extracts and outputs a high frequency component (100 Hz or more) of the input audio signal. The LPF 41 extracts and outputs a low frequency component (less than 100 Hz) of the input audio signal.

  The array speaker device 2 includes level adjusting units 70A to 70E in order to adjust the level ratio between the sound output from the woofers 33L and 33R and the sound output from the subwoofer unit 72.

  The level of each audio signal output from the LPF 41 is adjusted by the level adjusting units 70A to 70E. The audio signals whose levels are adjusted by the level adjusting units 70A to 70E are added by the adding unit 71, respectively. The audio signal added by the adder 71 is input to the subwoofer unit 72 via a DAC (not shown).

  Each audio signal output from the HPF 40 is input to the rear reverberation processing unit 44. The rear reverberation sound processing unit 44 superimposes the audio signal of the rear reverberation sound on each input audio signal and outputs it to the level adjustment unit 43 (43FL, 43FR, 43C, 43SR, 43SL).

  More specifically, the rear reverberation processing unit 44 includes a gain adjustment unit 441, a rear reverberation generation unit 442, and a synthesis unit 443, as shown in FIG. Each audio signal input to the rear reverberation processing unit 44 is input to the gain adjustment unit 441 and the synthesis unit 443. The gain adjusting unit 441 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 221 of the initial reflected sound processing unit 22. 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 442.

  The rear reverberation sound generation unit 442 generates an audio signal of the rear reverberation sound based on each input audio signal.

  As shown in FIG. 4, 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 442 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). For example, 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 222 (300 msec from the direct sound generation) has elapsed. Accordingly, the rear reverberation sound generation unit 442 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 443. Although the example in which the rear reverberation sound generation unit 442 is realized by an IIR filter has been shown, it can also be realized by using an FIR filter.

  Each audio signal output from the rear reverberation sound generation unit 442 is input to the synthesis unit 443. As shown in FIGS. 2 and 3B, the synthesis unit 443 synthesizes each audio signal input from the rear reverberation sound generation unit 442 with each audio signal input from the HPF 40, and combines each audio signal The signal is output to the level adjustment unit 43. Thereby, the rear reverberation sound is superimposed on the direct sound (corresponding to the audio signal directly input from the HPF 40 to the synthesis unit 443). In other words, the characteristic of the rear reverberant sound is added to the direct sound. The rear reverberation sound is output from the woofers 33L and 33R together with the sound that makes the virtual sound source be perceived.

  The level adjustment unit 43 adjusts the level of each input audio signal and outputs it to the localization adding unit 42 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 42 performs processing for localizing each input audio signal to a virtual sound source position. In order to localize an audio signal to a virtual sound source position, 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. By giving HRTF to the audio signal and emitting sound from the woofer 33L (or the woofer 33R), the listener perceives it as being emitted from the virtual speaker.

  As shown in FIG. 5A, the localization adding unit 42 includes filters 421L to 425L and filters 421R to 425R for convolving the HRTF impulse response for each channel.

  The FL channel audio signal (the audio signal output from the HPF 40FL) is input to the filters 421L and 421R. The filter 421L applies an HRTF of a path from the position of the virtual sound source VSFL (see FIG. 6) on the left front of the listener to the left ear to the FL channel audio signal. The filter 421R 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.

  The filters 423L to 425L 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 left ear of the listener to the audio signal of the C, SL, SR channel. The filters 423R to 425R 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.

  Then, the adding unit 426L synthesizes the audio signals output from the filters 421L to 425L, and outputs the synthesized audio signal to the crosstalk cancellation processing unit 50 as an audio signal VL. The adder 426R synthesizes the audio signals output from the filters 421R to 425R, and outputs the synthesized audio signal VR to the crosstalk cancellation processing unit 50.

  The crosstalk cancellation processing unit 50 cancels the crosstalk that is emitted from the woofer 33L and reaches the right ear, and the woofer 33L and the woofer 33R so that the direct sound that is emitted from the woofer 33L and reaches the left ear can be heard flat. Change the frequency characteristics of each input audio signal. Similarly, the crosstalk cancellation processing unit 50 cancels the crosstalk that is emitted from the woofer 33R and reaches the left ear, and so that the direct sound that is emitted from the woofer 33R and reaches the right ear can be heard flat. The frequency characteristic of each audio signal input to the woofer 33R is changed.

  More specifically, the crosstalk cancellation processing unit 50 performs processing using the correction unit 51 and the combining units 52L and 52R.

  As shown in FIG. 5B, the correction unit 51 includes direct correction units 511L and 511R and cross correction units 512L and 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 emitted from the woofer 33L is emitted near the left ear. In the direct correction unit 511L, a filter coefficient is set such that the sound output from the woofer 33L can be heard flat at the position of the left ear. The direct correction unit 511L corrects the input audio signal VL and outputs an audio signal VLD.

  The cross correction unit 512R, in combination with the synthesizing unit 52L, outputs from the woofer 33L a sound having a phase opposite to that from the woofer 33R to cancel the sound pressure at the position of the left ear, thereby canceling the sound of the woofer 33R. Is suppressed from being heard in the left ear. In addition, the cross correction unit 512R performs a process of making the listener perceive that the sound emitted from the woofer 33L is emitted near the left ear. The cross correction unit 512R is set with a filter coefficient that prevents the sound output from the woofer 33R from being heard at the position of the left ear. The cross correction unit 512R corrects the input audio signal VR and outputs an audio signal VRC.

  The synthesizer 52L synthesizes the audio signal VRC with the audio signal VLD in reverse phase.

  The direct correction unit 511R performs processing to make the listener perceive that the sound emitted from the woofer 33R is emitted near the right ear. In the direct correction unit 511R, a filter coefficient is set so that the sound output from the woofer 33R can be heard flat at the position of the right ear. The direct correction unit 511R corrects the input audio signal VR and outputs an audio signal VRD.

  The cross correction unit 512L, in combination with the synthesis unit 52R, outputs from the woofer 33R a sound having a phase opposite to that from the woofer 33L to cancel the sound pressure at the position of the right ear, thereby canceling the sound of the woofer 33L. Is suppressed from being heard in the right ear. In addition, the cross correction unit 512L performs a process of making the listener perceive that the sound emitted from the woofer 33R is emitted near the right ear. The cross correction unit 512L is set with a filter coefficient that prevents the sound output from the woofer 33L from being heard at the position of the right ear. The cross correction unit 512L corrects the input audio signal VL and outputs an audio signal VLC.

  The synthesizer 52R synthesizes the audio signal VLC with the audio signal VLC in reverse phase.

  The audio signal output from the synthesis unit 52L is input to the delay processing unit 60L. The audio signal is delayed by a predetermined time by the delay processing unit 60L and input to the level adjusting unit 61L. The audio signal output from the combining unit 52R is input to the delay processing unit 60R. The delay processing unit 60R delays the audio signal by the same delay time as the delay processing unit 60L.

  The delay times by the delay processing units 60L and 60R are 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 device 2 includes a delay processing unit for each channel after the directivity control unit 20 and delays the sound beam so that the sound beam does not disturb the sound that causes the virtual sound source to be perceived. It does not matter.

  The level adjusters 61L and 61R are provided to collectively adjust the sound levels that cause the virtual sound sources of all channels to be perceived. The level adjusters 61L and 61R adjust the level of each audio signal delayed by the delay processors 60L and 60R. The audio signals whose levels are adjusted by the level adjusters 61L and 61R are input to the woofers 33L and 33R via the adders 32L and 32R.

  Since the audio signals outside the audio beam band (less than 200 Hz) output from the speaker units 21A to 21P are input to the adders 32L and 32R, the sound outside the audio beam band and the sound that localizes the virtual sound source Is output by the woofers 33L and 33R.

  As described above, the array speaker device 2 localizes the audio signal of each channel on which the audio signal of the rear reverberation sound is superimposed at the virtual sound source position.

  Next, the sound field generated by the array speaker device 2 will be described with reference to FIG. In FIG. 6, the white arrow indicates the path of the sound beam output from the array speaker device 2, and the plurality of arcs indicate sound that perceives the virtual sound source output from the array speaker device 2. In FIG. 6, the asterisk indicates the position of the sound source generated by the sound beam and the position of the virtual sound source.

  As shown in FIG. 6, the array speaker device 2 outputs five sound beams according to the number of channels of the input audio signal. For example, the C channel audio signal is subjected to delay control such that the focal position is set behind the array speaker device 2. 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 subjected to delay control so that, for example, 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 SL and SR channel audio signals are subjected to delay control so that, for example, 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 421L to 425L and the filters 421R to 425R of the localization adding unit 42 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. 6, 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.

  Thereby, the array speaker apparatus 2 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.

  Here, as shown in FIG. 6, the array speaker device 2 superimposes the initial reflected sound on each sound beam. 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. In addition, 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.

  Further, the array speaker device 2 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 2 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. Thereby, the array speaker apparatus 2 prevents the sense of 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. .

  As described above, the array speaker device 2 imparts the sound field effect to the sound of the content by the initial reflection sound and the rear reverberation sound without impairing the effect of sound localization imparting each sound beam and the virtual sound source. Can do.

  Moreover, the array speaker apparatus 2 can change the level ratio of the initial reflection sound and the rear reverberation sound to a desired ratio of the listener by including a set of the gain adjustment unit 221 and the gain adjustment unit 441.

  Further, the array speaker device 2 outputs a sound beam and a sound that allows a virtual sound source to be perceived with respect to an audio signal of multi-channel surround sound, and gives a sound field effect. Therefore, the array speaker device 2 can impart a sound field effect to the sound of the content while giving a sense of localization so as to surround the listener.

  In the above example, the rear reverberation sound generated by the rear reverberation sound generation unit 442 is output from the woofers 33L and 33R after being superimposed on the sound causing the virtual sound source to be perceived. It does not have to be superimposed on. For example, the audio signal of the rear reverberation sound generated by the rear reverberation sound generation unit 442 may be input to the woofers 33L and 33R via the level adjustment units 34L and 34R without passing through the localization adding unit 42.

  Next, a speaker set 2A according to a modification of the array speaker device 2 will be described with reference to the drawings. FIG. 7 is a diagram for explaining the speaker set 2A. FIG. 8 is a part of a block diagram of the speaker set 2 </ b> A and the subwoofer 3. However, the arrows in FIG. 7 indicate sound paths having directivity in the interior 900 of the automobile.

  The speaker set 2A is different from the array speaker device 2 in that the directional speaker unit 21 (21Q, 21R, 21S, 21T, 21U) outputs sound having directivity. A description of the same configuration as that of the array speaker device 2 is omitted.

  Each directional speaker unit 21 is arranged according to a channel. That is, the directional speaker unit 21S corresponding to the C channel is disposed in front of the listener. The directional speaker unit 21Q corresponding to the FL channel is arranged in front of the listener and on the left side. The directional speaker unit 21R corresponding to the FR channel is disposed in front of and right of the listener. The directional speaker unit 21T corresponding to the SL channel is disposed behind and to the left of the listener. The directional speaker unit 21U corresponding to the SR channel is disposed behind and to the right of the listener.

  Each audio signal output from the level adjustment unit 18 is input to the delay processing unit 23 (23FL, 23FR, 23C, 23SR, 23SL) as shown in FIG. The delay processing unit 23 performs delay processing according to the path length from each directional speaker unit 21 to the listener so that the phase of sound having directivity is aligned in the vicinity of the listener.

  Each audio signal output from the delay processing unit 23 is input to each directional speaker unit 21. Even in such a configuration, the speaker set 2 </ b> A can superimpose the initial reflected sound on the sound having directivity corresponding to each channel and deliver it to the listener.

  In this modification, the delay times of the delay processing unit 60 and the delay processing unit 23 are set so that sound having directivity and sound that perceives a virtual sound source are not output at the same timing.

DESCRIPTION OF SYMBOLS 1 ... AV system 2, 2A ... Array speaker apparatus 3 ... Subwoofer 4 ... Television 10 ... Decoder 11 ... DIR
12 ... ADC
13 ... HDMI receiver 14FL, FR, C, SR, SL ... HPF
15FL, FR, C, SR, SL ... LPF
16, 17, addition unit 18, level adjustment unit 20, directivity control units 21 </ b> A to 21 </ b> P, speaker units 21 </ b> Q, 21 </ b> R, 21 </ b> S, 21 </ b> U, 21 </ b> T, directional speaker unit 22, initial reflected sound processing unit 221, gain adjustment unit 222. ... Initial reflection sound generator 223 ... Synthesizers 30L, 30R ... HPF
31L, 31R ... LPF
32L, 32R ... Adder 33L, 33R ... Woofer 40FL, FR, C, SR, SL ... HPF
41FL, FR, C, SR, SL ... LPF
42 ... Localization addition unit 43 ... Level adjustment unit 44 ... Rear reverberation sound processing unit 441 ... Gain adjustment unit 442 ... Rear reverberation sound generation unit 443 ... Synthesis unit 50 ... Crosstalk cancellation processing unit 51 ... Correction units 52L, 52R ... Synthesis unit 60L, 60R ... delay processing units 61L, 61R ... level adjusting units 70A to 70E, 70F, 70G ... level adjusting unit 71 ... adding unit 72 ... subwoofer unit

Claims (5)

An input section to which an audio signal is input;
A first sound emitting unit that emits sound based on the input audio signal;
A second sound emitting unit 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;
An initial reflected sound adding unit for adding characteristics of the initial reflected sound to the input audio signal;
A rear reverberation adding unit for adding a characteristic of the rear reverberation to the input audio signal;
With
The initial reflected sound adding unit adds the characteristics of the initial reflected sound to the audio signal input to the second sound emitting unit,
The rear reverberation sound adding unit adds the characteristics of the rear reverberant sound to the audio signal input to the first sound emission unit.
Speaker device. A level adjusting unit that adjusts a level ratio between the initial reflected sound of the initial reflected sound adding unit and the rear reverberant sound of the rear reverberant sound adding unit;
The speaker device according to claim 1. The audio signal is an audio signal of multi-channel surround sound.
The speaker device according to claim 1 or 2. The first sound emitting unit outputs sound having directivity.
The speaker device according to claim 1. The first sound emitting unit is
It consists of a stereo speaker to which the audio signal of the localization adding unit is input,
The second sound emitting unit is
A speaker array;
A directivity control unit that delays and distributes the audio signals input to the input unit to the speaker array, respectively.
The speaker device according to claim 4.

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