Spectrum Compensation Filter for Adjacent Sound Source Groups

Abstract

A method for generating signals to drive a first line array of audio sources, the first line array of audio sources comprising a primary audio source and one or more secondary audio sources. The method includes receiving an audio signal for a first channel of an audio system, deriving a first signal and a second signal from the audio signal, generating second drive signals to drive the one or more secondary audio sources by applying a low-pass filter to the second signal, and generating first drive signals to drive the primary audio sources by applying a corresponding high-frequency shelving filter to the first signal. A computer program product and an audio system for generating a flattened sound field are also provided.

Description

Technical Field

[0001] The present invention relates to a method for improving the spectral response of a plurality of coherent sound sources, where the delay between the sounds arriving at a reception point from the plurality of sound sources causes spectral variations across both frequency and space.

Background Art

[0002] The custom install (CI) market is growing in size for speaker manufacturers, and each manufacturer has experienced an increase in the number of their products being specified in new construction and remodeling projects. Many of these projects involve increasingly large spaces, such as home theaters that are 3 - 4 times the size of a normal living room. Even as the space gets larger, it is still desired to maintain a high sound pressure level (SPL) target value across the entire listening area. Furthermore, alongside this desire for high SPL target values, there is also a persistent desire for high-fidelity reproduction.

[0003] In the world of professional audio and live sound, there are several well-known solutions for achieving high SPL target values. For example, the concept of a line source array is well known, which approximates a line source using a closely arranged group of sound sources (drive unit group). Unlike conventional point source speakers, it attenuates by only -3 dB every time the distance doubles, rather than -6 dB every time the distance doubles. However, such an array requires a large number of drive units and complex mechanical design or computationally expensive processing to align the drive units to approximate the acoustic characteristics of the line source. Furthermore, in reality, even when using a line array, it is only possible to approximate a line source at low and mid frequencies. Therefore, in order to output a high SPL over a wide frequency range, alternative sound sources such as horn-loaded compression drivers must be used, which can achieve a high SPL but cannot achieve the high fidelity required in CI applications.

[0004] As an alternative, multiple high-fidelity speakers into which the same audio signal is input may be used as a single channel. FIG. 1 shows an exemplary CI installation as a home theater system 100, where three sets of three in-wall speakers are used behind and on either side of a projection screen 102 for each of the left, center, and right channels. A first set of three in-wall speakers 104 is behind the projection screen 102, a second set 106 is on the left side of the projection screen 102, and a third set 108 (composed of speakers 108a, 108b, and 108c) is on the right side of the projection screen 102. The same signal is input to each set of the three speakers. Depending on the desired SPL, the number of speakers used for each channel can be reduced or increased (the SPL increases by +6 dB for every doubling of the number of speakers). However, using multiple speakers to which the same signal is applied causes problems due to destructive interference, also known as comb filtering, between the multiple coherent sound sources.

[0005] This problem is known in a 2.5-way speaker consisting of three drive units, where one of the drive units operates in the highest frequency band and the other two of the drive units usually operate over slightly different frequency bands. One of the two identical drive units covers the frequency band up to the crossover with the highest frequency drive unit, and the other drive unit is low-pass filtered to supply additional low-frequency energy, so that the "baffle step" phenomenon can be overcome without causing interference in the mid-frequency band where the distance between the drive units causes comb filtering. However, the 2.5-way speaker still has performance problems.

[0006] Methods based on time delay, phase change, and beam steering can reduce or eliminate interference, but only for a given single point in space, and may actually increase interference at other positions.

[0007] Therefore, there is a need for an improved method for reducing interference between multiple coherent sound sources while maintaining the overall spectral balance.

Summary of the Invention

[0008] According to a first aspect of the present invention, there is provided a method for generating a signal for driving a first line array of sound sources, the first line array of sound sources comprising a primary sound source and one or more secondary sound sources. The method comprises receiving an audio signal for a first channel of an audio system, deriving a first signal and a second signal from the audio signal, generating a second drive signal for driving the one or more secondary sound sources by applying a low-pass filter to the second signal, and generating a first drive signal for driving the primary sound source by applying a corresponding high-frequency shelving filter to the first signal including the steps of. In this way, interference between multiple coherent sound sources can be reduced while maintaining the overall spectral balance.

[0009] According to a second aspect of the present invention, there is provided a computer program product comprising computer-executable code that, when executed on one or more processors of an audio system, causes the system to perform the method of the first aspect. In this way, while the method of the first aspect of the present invention can be implemented by one or more processors of an audio system to reduce interference between a plurality of coherent sound sources, the overall spectral balance can be maintained. To implement the method using one or more processors, the method may be executed by a single processor of the audio system or may be executed across multiple processors.

[0010] According to a third aspect of the present invention, an audio system comprises a group of one or more digital signal processors configured to perform the method described above. In this way, the audio system can implement the method described above using only one or more digital signal processors.

[0011] According to a fourth aspect of the present invention, an audio system for generating a flattened sound field comprises a first line array of sound sources comprising a primary sound source and one or more secondary sound sources. The primary sound source is driven by a first drive signal and the secondary sound source is driven by a second drive signal. The first signal and the second signal are derived from an audio signal received for a first channel of the audio system. A low-pass filter is applied to the second signal to generate the second drive signal, and a corresponding high-frequency shelving filter is applied to the first signal to generate the first drive signal. In this way, interference between a plurality of coherent sound sources can be reduced while maintaining the overall spectral balance.

[0012] Preferably, the method further comprises applying an all-pass filter to the first signal. In this way, compensation is made for additional interference caused by the relative phase response of the low-pass filter and the high-frequency shelving filter that results in energy loss near the characteristic frequency of the filter.

[0013] Optionally, the method further includes applying additional, different all-pass filters to the first signal and the second signal. In this way, the time alignment between the first drive signal and the second drive signal is improved.

[0014] In one embodiment, each of the characteristic frequencies of the low-pass filter and the high-frequency shelving filter is approximately equal to the reciprocal of twice the time delay between when sound reaches the listening position from the primary sound source and the one or more secondary sound sources. In this way, the characteristic frequencies of each filter bank are at the frequencies at which a first notch of destructive interference occurs between at least two sound sources. This ensures that while the filter maximally reduces interference between multiple coherent sound sources, the overall spectral balance is maintained.

[0015] In one embodiment, the gain g of the high-frequency shelving filter is g = 20 log10(N + 1), where N is the number of secondary sound sources. This ensures that the high-frequency shelving filter is applied in an approximately accurate manner and maximally reduces interference between multiple coherent sound sources while maintaining the overall spectral balance.

[0016] Optionally, the first line array of sound sources may be a first line array of speakers including a primary speaker and one or more secondary speakers.

[0017] The computer program product of the second aspect of the present invention can be implemented as an update or improvement to an existing digital signal processor sound source system, or alternatively, as an update or improvement to an existing multi-channel or stereo audio processor. Thereby, an existing system can be updated by providing an update to an existing audio system.

[0018] Preferably, in the present audio system, the high-frequency shelving filter is implemented by a digital signal processor associated with the primary sound source, and the low-pass filter is implemented by at least one digital signal processor associated with the one or more secondary sound sources. Thus, filtering can be performed at the removed level to supply the first drive signal and the second drive signal to the primary sound source and the one or more secondary sound sources. Alternatively, filtering may be performed in a local digital signal processor within the audio system or in a digital signal processor within the drive unit of the sound source itself. However, in the case of a local digital signal processor and a digital signal processor within the drive unit, the digital signal processor is associated with the primary sound source or the one or more secondary sound sources, and thus, an appropriate filter is implemented to generate the corresponding first drive signal and second drive signal.

[0019] Preferably, the audio system may be a wall-mounted audio system. Thereby, depending on the positioning of the speakers and the proximity of the walls and other surfaces that reflect sound, it would be possible to minimize the reflection of sound from the walls that can cause destructive interference in an unpredictable way behind and around the sound source.

[0020] Optionally, the audio system may have a sound source consisting of a first line array of sound sources arranged vertically or horizontally. Thereby, the sound sources can be arranged optimally depending on the location where the audio system is installed.

[0021] In one embodiment, the audio system further comprises a second line array of sound sources driven by third and fourth drive signals derived from a second channel for the audio system in a similar manner as the first drive signal and the second drive signal and filtered in a similar manner as the corresponding signal groups in the first channel. Thereby, the concept of the present invention can be extended to two audio systems.

[0022] In one embodiment, the audio system may further comprise at least one line array of sound sources driven by a drive signal, which is derived from at least one further channel for the audio system in a similar manner to the first drive signal and the second drive signal and filtered in a similar manner to the corresponding signal group in the first channel. Thereby, the concept of the present invention can be extended to three or more channels of an audio system.

[0023] In one embodiment, the first line array of sound sources is a first line array of speakers including a primary speaker and one or more secondary speakers. Preferably, the audio system may have the first line array of speakers arranged such that the distance between the acoustic centers of each successive speaker of the first line array of speakers is between 15 cm and 30 cm. Thereby, the time delay between the sounds reaching the listening position from the primary speaker and the secondary speakers can be calculated, and then the frequency at which the first notch will occur can be calculated, and the characteristic frequencies at which a low-pass filter and a high-frequency shelving filter should be set can be accurately calculated.

[0024] As will be understood by those skilled in the art, the present invention can be realized in various ways depending on the application.

Brief Description of the Drawings

[0025] Examples of the present invention are described in detail with reference to the accompanying drawings.

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Embodiments for Carrying Out the Invention

[0026] The present invention can be realized in several different ways depending on the audio system being used. The following description will describe several examples with reference to the figures.

[0027] The present invention aims to mitigate the influence of spatial aliasing between two or more adjacent sound sources. The present invention is necessary when the groups of sound source signals for each adjacent sound source are coherent, as when using multiple speakers as a single channel within a home theater system 100 as shown in FIG. 1.

[0028] In the exemplary system shown in FIG. 1, the speaker group is arranged vertically, but it may be arranged horizontally. Further, the central set of speakers 104 is not essential, and the system may be a stereo system consisting only of the left set 106 and the right set 108 of speakers, or in fact, the system may be monophonic, consisting of only one set of speaker groups. The right set of speakers 108 is composed of speakers 108a, 108b, and 108c. One of these becomes the first speaker and two become the second speakers. Further, although the sound source in this example is a two-way wall-embedded speaker, the present invention can be applied to any adjacent coherent sound source.

[0029] To explain the problem to be overcome by the present invention, consider the system 200 given in FIG. 2. FIG. 2 shows a simple example of two sound sources 202 and 204 with a distance d1 meters between the acoustic centers. The listening position 206 marked with "X" is d2 meters away from one of the sound sources, i.e., the primary sound source 202, and is axially arranged with respect to this sound source in both the horizontal and vertical directions. From the Pythagorean theorem, the distance d3 to the other sound source, the secondary sound source 204, is d3 = √(d1 2 + d2 2 ) which is clearly greater than d2. Therefore, this causes a time delay of Δt (d3 - d2) / c seconds between the two sounds reaching the listening position 206 from the primary sound source 202 and from the secondary sound source 204. Here, c = 343 m / s is the speed of sound in air at 20 degrees Celsius. As a result, due to the destructive interference between the primary sound source 202 and the secondary sound source 204, a series of notch groups occur in the frequency response observed at the listening position 206. This is known as the comb filter effect. The notches occur at frequencies f n =(n / 2Δt)H Z where n are all odd integers.

[0030] This comb filter effect is illustrated in Figure 3, which plots the sound pressure level against frequency as compared to a single sound source. The "comb-shaped" notch group shown in Figure 3 is the destructive interference that occurs between two sound sources 202 and 204. The first notch 302 occurs at f1, the second notch 304 occurs at f3, the third notch 306 occurs at f5, and so on.

[0031] For example, if there is a distance of 50 centimeters between the primary sound source 202 and the secondary sound source 204, and the listening position 206 is 2 meters in front of the primary sound source 202, there will be a difference in propagation path length of 6.15 centimeters. This corresponds to a time delay of 179 microseconds between the two sounds reaching the listening position. Therefore, as shown in Figure 3, the frequency spectrum at the listening position has notches at odd multiples of f1 = 2.8 kHz.

[0032] This example consists of only two sound sources 202 and 204, but the principle is the same for any number of sound sources greater than two. However, the pattern of notches in the frequency response becomes more complex, and the notches appear at groups of frequencies corresponding to the time delays for each secondary sound source and at odd harmonics of these frequencies.

[0033] When the primary and secondary sound sources are speakers, the distance d1 between the acoustic centers of the sound sources can typically be between 15 cm and 30 cm. When the primary and secondary sound sources are drive units within a single speaker, the distance d1 between their acoustic centers may only be 5 cm. The farther apart the acoustic centers of the sound sources are, the more the comb filter effect extends to lower frequencies, thus losing headroom in the input signal for a high-frequency shelving filter. However, the upper limit of the distance d1 between the acoustic centers of the sound sources depends on the listening distance d2, and the greater the listening distance, the farther apart the sound source group can be from each other.

[0034] To reduce the comb filter effect, the present invention applies a low-pass filter to the secondary sound source 204 so that only the primary sound source 202 operates in the frequency group where destructive interference occurs. However, this effectively results in having one sound source above the low-pass and two sound sources below the low-pass, leading to a mismatch in SPL at frequencies above and below the low-pass (above and below f1).

[0035] Fortunately, as shown in FIG. 4, in music content above 1 kHz, the energy generally decreases with frequency, as shown by different datasets from Stuart, J.R. (2006) "Active loudspeakers", Proceedings of the 21st AES UK Conference: Audio at Home. Dataset IEC268-1 is the IEC standard noise spectrum for power testing of audio products, dataset Sivian and Adams is related to previous research, and dataset JRS is the data analysis conducted by the author of the paper. Therefore, in all four different datasets, there is a general decrease in energy above 1 kHz and below 100 Hz, so it is clear that this decrease in energy above 1 kHz occurs commonly in music content. By the decrease in energy at these high frequencies, potential headroom for processing is provided to compensate for the fact that there is only one contributing sound source above the cut-off frequency of the aforementioned low-pass filter. To achieve this compensation, a corresponding high-frequency shelving filter is applied to the primary sound source 202.

[0036] The gain of the high-frequency shelving filter is g = 20log 10(N + 1) law, depending on the number of secondary sound sources, where g is the gain of the shelving filter expressed in decibels and N is the number of secondary sound sources. Figure 5 illustrates the possible responses of the low-pass filter and the high-frequency shelving filter for N = 1 and N = 2 secondary sound sources. The solid line in Figure 5 illustrates the possible response of the high-frequency shelving filter for N = 1, the dashed line illustrates the possible response of the high-frequency shelving filter for N = 2, and the dotted line illustrates the possible response of the low-pass filter.

[0037] Figure 6 typically shows that both the low-pass filter 604 and the high-frequency shelving filter 602 will have a characteristic transition frequency, and they may be close to each other, but not necessarily the same as f1, and will be within or in the vicinity of the narrow frequency bandwidth of f1, which is the frequency of the first notch. The characteristic frequency groups of both the low-pass filter(s) and the high-frequency shelving filter can be predicted by the previously calculated f1608. As shown in Figure 6, typically, the characteristic frequency f of the high-frequency shelving filter 606 C1 is located slightly below the frequency of f1608, and the characteristic frequency f of the low-pass filter(s) 610 C2 will be located slightly above the frequency of f1608. However, the exact frequencies will need to be adjusted by those skilled in the art based on the specific system and implementation.

[0038] As shown in FIG. 4, the peak level of the frequency in music drops rapidly above 1 kHz, providing headroom for applying a high-frequency shelving filter, because in most real-world systems, destructive interference rarely appears below 1 kHz. That being said, in the case of non-typical signals (groups), sufficient care must be taken to ensure that the system provides appropriate protection to prevent damage to the sound source.

[0039] Accordingly, the present invention relates to a method of utilizing this headroom to reduce interference between multiple coherent sound sources while maintaining the balance of the overall spectrum.

[0040] FIG. 7 illustrates such an embodiment for three sound sources, namely one primary sound source 710 and two secondary sound sources 712 and 714. FIG. 7 shows that the audio signal 702 for the channel of the audio system is split at 704 to become the drive signal for the primary sound source 710 and the drive signals for the two secondary sound sources 712 and 714. The high-frequency shelving filter 706 is applied to the drive signal of the primary sound source 710, and the low-pass filter 708 is applied to the drive signals of the secondary sound sources 712 and 714.

[0041] A further embodiment shown in FIG. 8 introduces an all-pass filter 816 to the primary sound source 810. FIG. 8 shows that an audio signal 802 for a channel of an audio system is split at 804 into a drive signal for the primary sound source 810 and drive signals for two secondary sound sources 812 and 814. A high-frequency shelving filter 806 and an all-pass filter 816 are applied to the drive signal for the primary sound source 810, and a low-pass filter 808 is applied to the drive signals for the secondary sound sources 812 and 814. The all-pass filter 816 newly introduced to the primary sound source 810 is for compensating the phase shift of the low-pass filter 808 with respect to the secondary sound sources 812 and 814. For example, the second-order low-pass filter 808 results in a 180-degree phase shift about the center frequency of the filter. Thus, the primary all-pass filter 816 can be applied to the primary sound source 810 to apply a complementary 180-degree phase shift. Therefore, the center frequency of the all-pass filter 816 should approximate the center frequency used for the low-pass filter 808.

[0042] As shown in FIG. 9, a third preferred embodiment introduces additional all-pass filters 918 and 920 to both the primary sound source 910 and the secondary sound sources 912 and 914. FIG. 9 shows that the audio signal 902 for the channel of the audio system is split at 904 into a drive signal for the primary sound source 910 and drive signals for the two secondary sound sources 912 and 914. The high-frequency shelving filter 906, the all-pass filter 916, and the additional all-pass filter 918 are applied to the drive signal of the primary sound source 910, and the low-pass filter 908 and the all-pass filter 920 are applied to the drive signals of the secondary sound sources 912 and 914. The newly introduced all-pass filters 918 and 920 can be used to improve the time alignment between the primary drive signal and the secondary drive signals, and as a result, reduce the frequency cancellation effect of the comb filter. For example, in order to reduce the cancellation at the frequency of the first notch by inverting the phase relationship, the all-pass filter for the secondary sound source may be applied below the frequency (f1) of the first notch, and the all-pass filter for the primary sound source may be applied above the frequency (f1) of the first notch.

[0043] Figure 10 shows the simulated frequency responses at the listening position when not using the filter proposed by the present invention and when using different combinations of the filters proposed above. The dotted line 1002 shows the frequency response when no filter is applied. The dashed-dotted line 1004 shows the frequency response when only the low-pass filter and the high-frequency shelving filter are applied (as shown in FIG. 7). The dashed line 1006 shows the frequency response when an all-pass filter is applied to the primary sound source in addition to the low-pass filter and the high-frequency shelving filter (as shown in FIG. 8). The solid line 1008 shows the frequency response when an additional all-pass filter is added to both the primary sound source and the secondary sound source in addition to all the other filters applied (as shown in FIG. 9). It will be appreciated that all combinations of the proposed filters greatly reduce spectral variations. However, it will be appreciated that when an additional all-pass filter is applied, the spectral variations are further reduced compared to other combinations of filters.

[0044] In addition, as shown in FIGS. 11A and 11B, the proposed invention not only improves the frequency response at the listening position but also reduces spectral variations across the location. FIG. 11A shows the variations in sound pressure level across the location when no filter is applied. FIG. 11B shows the variations in sound pressure level across the location when all filters are applied as shown in FIG. 9. The horizontal axes 1102 in both FIGS. 11A and 11B represent the distance off-axis of the listening position in the plane of the sound source array. The vertical axis 1104 represents the distance from the array of listening positions. The contour lines within the plot represent the SPL at that position in decibels, and each line represents a 3-decibel (dB) decrease in SPL. Some contour lines representing decreases in multiples of 6 dB are labeled accordingly.

[0045] As can be seen from FIG. 11A, when the filter is not applied, there is significant destructive interference as shown by the change in the contour lines, and the high SPL regions are labeled 1110. Conversely, in FIG. 11B, when filtering is applied as shown in FIG. 9, there is no change in the contour lines and the SPL decreases uniformly.

[0046] In a preferred embodiment, the low-pass filter, the high-frequency shelving filter, and the all-pass filter are 2-pole 2-zero digital biquad filters, the design of which is known to those skilled in the art. Such a group of filters is preferred because of the fact that examples of implementing these groups of filters are simple, computationally efficient, and supported in many existing signal processing systems. However, more complex designs of the filter can be used, and the group of filters can be implemented in either the analog or digital domain, and by either software or hardware.

[0047] In certain embodiments, the filter can be implemented as an update or improvement to an existing system, or as part of the design of a new system. Further, in certain embodiments, the filter may be implemented internally within the system, such as within each speaker shown in FIG. 1 for example, while in other embodiments the filter will be applied externally in a pre-processor device.

[0048] An odd number of sound sources is preferred to maintain symmetry in the radiated sound field. Further, to maximize the effectiveness of the filter and suppress the required gain of the shelving filter, the preferred number of sound sources is 3. However, the present invention may be applied to any number of adjacent sound source groups greater than 1.

Claims

A method for generating a signal for driving a first line array of a plurality of coherent sound sources, wherein the first line array of the plurality of coherent sound sources comprises a primary sound source and one or more secondary sound sources, the method comprising: Receiving an audio signal for a first channel of an audio system; Deriving from the audio signal a first drive signal for driving the primary sound source and a second drive signal for driving the one or more secondary sound sources; Applying a low-pass filter to the second drive signal for driving the one or more secondary sound sources, wherein the low-pass filter reduces energy at high frequencies so as to suppress destructive interference occurring between the sound from the primary sound source and the sound from the one or more secondary sound sources; Applying; and Applying a corresponding high-frequency shelving filter to the first drive signal for driving the primary sound source, wherein the high-frequency shelving filter compensates for the reduction of energy at high frequencies by the low-pass filter, and the gain g of the high-frequency shelving filter is g = 20 log10 (N + 1), where N is the number of the secondary sound sources; Applying A method comprising the steps of. Claim 2 Further comprising applying a first all-pass filter to the first drive signal so as to compensate for additional interference caused by the relative phase responses of the low-pass filter and the high-frequency shelving filter that result in energy loss in the vicinity of the characteristic transition frequency of the filter The method according to claim 1. Claim 3 Further comprising applying a second all-pass filter to the first drive signal and applying a third all-pass filter to the second drive signal so as to improve the time alignment between the first drive signal and the second drive signal The method according to claim 1 or 2.

4. The characteristic transition frequency of each of the low-pass filter and the high-frequency shelving filter is equal to the reciprocal of twice the time delay for sound to reach the listening position from the primary sound source and the one or more secondary sound sources. The method according to any one of claims 1 - 3.

5. The line array of the plurality of coherent sound sources is a first line array of speakers including a primary speaker and one or more secondary speakers. The method according to any one of claims 1 - 4.

6. A computer program product comprising computer-executable code that causes the audio system to execute the method according to any one of claims 1 - 5 when executed on one or more processors of the audio system.

7. It is realized as an update or improvement of the sound source system of an existing digital signal processor. The computer program product according to claim 6.

8. It is realized as an update or improvement of an existing multi-channel or stereo audio processor. The computer program product according to claim 6.

9. An audio system comprising one or more groups of digital signal processors configured to execute the method according to any one of claims 1 - 5.

10. The high-frequency shelving filter is realized by a digital signal processor associated with the primary sound source, and the low-pass filter is realized by at least one digital signal processor associated with the one or more secondary sound sources. The audio system according to claim 9.

11. An audio system for generating a flattened sound field, the audio system comprising: A first line array of a plurality of coherent sound sources including a primary sound source and one or more secondary sound sources, The primary sound source is driven by a first drive signal, and the one or more secondary sound sources are driven by a second drive signal, The first drive signal and the second drive signal are derived from an audio signal received for a first channel of the audio system, A first line array; A low-pass filter applied to the second drive signal, The low-pass filter reduces energy at high frequencies so as to suppress destructive interference occurring between the sound from the primary sound source and the sound from the one or more secondary sound sources. A low-pass filter; A high-frequency shelving filter applied to the first drive signal, The high-frequency shelving filter compensates for the reduction in energy at high frequencies by the low-pass filter, and the gain g of the high-frequency shelving filter is g = 20 log10 (N + 1), where N is the number of secondary sound sources. A high-frequency shelving filter; An audio system comprising.

12. Further comprising a first all-pass filter applied to the first drive signal so as to compensate for additional interference caused by the relative phase responses of the low-pass filter and the high-frequency shelving filter that result in energy loss in the vicinity of the characteristic transition frequency of the filter. The audio system according to claim 11.

13. Further comprising additional respective second all-pass filter and third all-pass filter applied to both the first drive signal and the second drive signal so as to improve the time alignment between the first drive signal and the second drive signal. The audio system according to claim 11 or 12.

14. The characteristic transition frequency of each of the low-pass filter and the high-frequency shelving filter is equal to the reciprocal of twice the time delay during which sound reaches the listening position from the primary sound source and the one or more secondary sound sources. The audio system according to any one of claims 11 - 13.

15. The plurality of coherent sound sources of the first line array are for attachment to a wall surface. The audio system according to any one of claims 11 - 14.

16. The first line array of the plurality of coherent sound sources is arranged vertically or horizontally. The audio system according to any one of claims 11 - 15.

17. In a manner similar to the first drive signal and the second drive signal, derived from an audio signal received for a second channel of the audio system and filtered in a manner similar to the corresponding signal group in the first channel, further comprising a second line array of a plurality of coherent sound sources driven by a third drive signal and a fourth drive signal. The audio system according to any one of claims 11 - 16.

18. In a manner similar to the first drive signal and the second drive signal, derived from an audio signal received for at least one further channel of the audio system and filtered in a manner similar to the corresponding signal group in the first channel, further comprising at least one further line array of a plurality of coherent sound sources including a primary sound source and one or more secondary sound sources driven by a fifth drive signal and a sixth drive signal. The audio system according to claim 17.

19. The first line array of the plurality of coherent sound sources is a first line array of a plurality of coherent speakers including a primary speaker and one or more secondary speakers. The audio system according to any one of claims 11 - 18.

20. The first line array of the plurality of coherent speakers is arranged such that the distance between the acoustic centers of the respective next speakers of the first line array of the plurality of coherent speakers is between 15 cm and 30 cm. The audio system according to claim 19.

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