Method for generating a signal for driving a first line array of a plurality of coherent sound sources and an audio system for generating a flat sound field

Abstract

A method of generating a signal for driving a first line array of sound sources. The first line array of sound sources includes a main sound source and one or more secondary sound sources. The method includes the steps of receiving an audio signal for a first channel of an audio system, deriving a first signal and a second signal from the audio signal, applying a low pass filter to the second signal to generate a second drive signal for driving the one or more secondary sound sources, and applying a corresponding high frequency shelf filter to the first signal to generate a first drive signal for driving the main sound source. A computer program product and an audio system for generating a flat sound field are also provided.

Description

Technical Field

[0001] The present invention relates to a method for improving the spectral response of multiple coherent sound sources, wherein the delays between the sounds arriving at a receiving point from the sound sources result in spectral variations in frequency and space. Background Art

[0002] The Custom Install (CI) market is growing in size for loudspeaker manufacturers, who are seeing their products increasingly specified in new homes and renovations. Many of these projects consist of larger and larger spaces; for example, large home theaters are 3 to 4 times the size of an average living room. With these larger spaces, there is still a desire to maintain high sound pressure level (SPL) targets throughout the listening area. Furthermore, despite these demands for high SPL targets, there is also a demand for high-fidelity playback.

[0003] In professional audio and live sound, there are well-known solutions for generating high SPL levels. For example, the concept of line source arrays is well known; wherein close sources (drive units) are used to approximate line sources, with a -3dB attenuation for each doubling of distance, rather than -6dB for each doubling as with conventional point source loudspeakers. However, such arrays require a large number of drive units, or require complex mechanical designs, or require computationally expensive processing to align the drive units to approximate the line source acoustic characteristics. In addition, in practice, line arrays can only approximate line sources at low and mid frequencies. Therefore, alternative sound sources such as horn-loaded compression drivers must be used to provide high SPL at high frequencies, but when providing high SPL, they cannot provide the high fidelity required in CI applications.

[0004] Another option is to use multiple high-fidelity speakers fed the same audio signal as a single channel. Figure 1 An exemplary CI device is shown as a home theater system 100 having three groups of 3 in-wall speakers used behind and on both sides of a projection screen 102, for each of the left, center, and right channels. A first group of 3 in-wall speakers 104 is behind the projection screen 102, a second group 106 is located to the left of the projection screen 102, and a third group 108 (consisting of speakers 108a, 108b, and 108c) is to the right of the projection screen 102. Each group of 3 speakers is fed the same signal. Depending on the desired SPL (each doubling of the number of speakers results in a +6dB increase in SPL), fewer or more speakers can be used for each channel. However, using multiple speakers fed the same signal can cause problems due to destructive interference between multiple coherent sound sources, sometimes called comb filtering.

[0005] This problem is known in 2.5-way loudspeakers, which consist of three drivers, one of which operates in the highest frequency range, while the other two drivers are usually identical, but operate in slightly different frequency ranges. One of the two identical drivers covers the frequency range until the crossover with the highest frequency driver, while the other is low-pass filtered to provide additional low frequency energy to overcome the "baffle step" phenomenon without introducing interference in the middle range where the distance between the drivers may introduce comb filtering. However, 2.5-way loudspeakers still have performance problems.

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

[0007] Therefore, improved methods are needed to reduce interference between multiple coherent sources while maintaining overall spectral balance. Summary of the invention

[0008] According to a first aspect of the invention, there is provided a method for generating a signal for driving a first line array of sound sources, wherein the first line array of sound sources comprises a main sound source and one or more secondary sound sources. The method comprises the following steps: receiving an audio signal for a first channel of an audio system, deriving a first signal and a second signal from the audio signal, applying a low pass filter to the second signal to generate a second drive signal for driving one or more secondary sound sources, and applying a corresponding high frequency shelf filter to the first signal to generate a first drive signal for driving the main sound source. This can reduce interference between multiple coherent sources while maintaining overall spectral balance.

[0009] According to a second aspect of the invention, a computer program product comprises computer executable code which, when executed on one or more processors of an audio system, causes the system to perform the method of the first aspect. Thus, the method of the first aspect of the invention can be implemented by one or more processors of an audio system to reduce interference between multiple coherent sources while maintaining overall spectral balance. By implementing the method with one or more processors, the method can be performed by a single processor of the audio system, or can be performed across multiple processors.

[0010] According to a third aspect of the present invention, the audio system comprises one or more digital signal processors adapted to perform the above method. Thus, the audio system can implement the above method using only one or more digital signal processors.

[0011] According to a fourth aspect of the invention, an audio system for generating a flat sound field comprises a first line array of sound sources, the sound sources comprising a main sound source and one or more auxiliary sound sources. The main sound source is driven by a first drive signal, and the auxiliary sound source is driven by a second drive signal. A first signal and a second signal are derived from a received audio signal for a first channel of the audio system. A low-pass filter is applied to the second signal to generate a second drive signal, and a corresponding high-frequency shelf filter is applied to the first signal to generate a first drive signal. This can reduce interference between multiple coherent sources while maintaining overall spectral balance.

[0012] Preferably, the method further comprises applying an all-pass filter to the first signal. In this way, additional interference introduced by the relative phase response of the low-pass and high-shelf filters, which results in energy loss around the characteristic frequencies of the filters, is compensated.

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

[0014] In some embodiments, the characteristic frequency of each of the low pass filter and the high frequency shelf filter is approximately the inverse of twice the time delay between the sound arriving at the listening position from the primary sound source and the one or more secondary sound sources. In this way, the characteristic frequency of each filter is at the frequency of the first notch of destructive interference occurring between at least two sound sources. This ensures that the filter has the maximum effect of reducing interference between multiple coherent sources while maintaining overall spectral balance.

[0015] In some embodiments, the gain of the high frequency shelf filter is g=20log 10 (N+1), where N is the number of secondary sources. This ensures that the high frequency shelf filter is applied in an appropriate manner to ensure maximum effect in reducing interference between multiple coherent sources while maintaining overall spectral balance.

[0016] Optionally, the first line array of sound sources may be a first line array of speakers comprising a main 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 enhancement to an existing digital signal processor sound source system, or as an update or enhancement to an existing multi-channel or stereo audio processor. In this way, an existing system can be updated by providing an update to an existing audio system.

[0018] Preferably, in the audio system, the high frequency shelf 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 one or more secondary sound sources. This allows filtering to be performed at different levels to provide the first drive signal and the second drive signal to the primary sound source and the one or more secondary sound sources. Alternatively, the filtering can be performed by a local digital signal processor within the audio system or by 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, so that appropriate filters are implemented for generating the corresponding first and second drive signals.

[0019] Preferably, the audio system may be an in-wall audio system. This ensures minimal sound reflections from walls, which may cause destructive interference behind and around the sound source in unpredictable ways, depending on the location of the speakers and their proximity to walls and other sound-reflecting surfaces.

[0020] Alternatively, the audio system may have a first line array of sound sources arranged vertically or horizontally. In this way, the sound sources may be located in the best position for the audio system installation.

[0021] In some embodiments, the audio system may further include a second line array of sound sources driven by third and fourth drive signals, the third and fourth drive signals being derived from a second channel for the audio system in the same manner as the first and second drive signals, and filtered in the same manner as the corresponding signals in the first channel. In this way, the concepts of the present invention may be extended to two audio systems.

[0022] In some embodiments, the audio system may further comprise at least one further line array of sound sources driven by drive signals for at least one further channel of the audio system derived in the same manner as the first drive signal and the second drive signal and filtered in the same manner as the corresponding signals in the first channel. In this way, the concepts of the present invention may be extended to three or more channels of an audio system.

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

[0024] As understood by those skilled in the art, the present invention can be implemented in various embodiments according to the present application. BRIEF DESCRIPTION OF THE DRAWINGS

[0025] Embodiments of the present invention will be described in detail below with reference to the accompanying drawings, in which:

[0026] Figure 1 An example installation of multiple in-wall speakers is shown.

[0027] Figure 2 Two sound sources are shown to illustrate an example of how time delay occurs.

[0028] Figure 3 Shows Figure 2 An example of comb filtering in the frequency response of the system shown.

[0029] Figure 4 An example of a power spectrum of typical well-known music is shown.

[0030] Figure 5 Possible filter responses for different numbers of secondary sound sources are shown.

[0031] Figure 6 A typical relationship between the characteristic frequencies of the low pass and high shelf filters and the first notch frequency of the comb filter in the present invention is shown.

[0032] Figure 7 A schematic diagram of an embodiment implementing the present invention for three sound sources is shown.

[0033] Figure 8 A schematic diagram of a second embodiment implementing the present invention for three sound sources is shown.

[0034] Fig. 9 A schematic diagram of a third preferred embodiment implementing the present invention for three sound sources is shown.

[0035] Fig.10 Examples of how processing in three different embodiments changes the sound pressure level relative to a single sound source are shown.

[0036] Fig.11A is a contour plot showing the spectral variation over space without the proposed filter.

[0037] Fig. 11B is a contour plot showing the spectral variation over space when the proposed filter is utilized. DETAILED DESCRIPTION

[0038] The invention can be implemented in many different ways depending on the audio system used. Some example implementations are described below with reference to the accompanying drawings.

[0039] The present invention aims to mitigate the effects of spatial aliasing between two or more very close sound sources. Figure 1 As shown, the present invention is necessary when multiple speakers are used as a single channel in the home theater system 100.

[0040] Despite Figure 1 In the example system in FIG. 1 , the speakers are mounted vertically; they may also be mounted horizontally. Furthermore, a center speaker set 104 is not required, and the system may be a stereo system consisting of only the left 106 and right 108 sets of speakers, or indeed the system may be mono and consist of only one set of speakers. The right set of speakers 108 consists of speakers 108a, 108b, and 108c. One of these will be the primary speaker and two will be secondary speakers. Additionally, while the sound sources in this example are two-way in-wall speakers, the present invention may be applied to any closely located coherent sound source.

[0041] To illustrate the problem that the present invention seeks to overcome, consider Figure 2 The system 200 is given in FIG. Figure 2 A simple example of two sound sources 202 and 204 with a distance d1 meters between their acoustic centers is shown. The listening position 206 marked by an 'X' is d2 meters away from one of the sound sources (i.e., the primary sound source 202) and is located on the same axis horizontally and vertically with respect to that sound source. According to the Pythagorean theorem, the distance to the other sound source (i.e., the secondary sound source 204) is This results in a time delay between the sound arriving at the listening position 206 from the primary sound source 202 and the secondary sound source 204. seconds, where c = 343 m / s is the speed of sound in air at 20 degrees Celsius. This results in a series of notches in the frequency response observed at the listening position 206 due to destructive interference between the primary sound source 202 and the secondary sound source 204. This is known as "comb filtering". The notches will be at the frequencies Hz, where n is any odd integer.

[0042] Figure 3 This comb filtering effect is shown in , which plots frequency versus sound pressure level relative to a single sound source. Figure 3 The "comb" notches shown are destructive interference that occurs between two sound sources 202 and 204. The first notch 302 is located at fl, the second notch 304 is located at f3, the third notch 306 is located at f5, and so on.

[0043] For example, the distance between the primary sound source 202 and the secondary sound source 204 is 50 cm, and the listening position 206 is 2 meters in front of the primary sound source 202, resulting in a path length difference of 6.15 cm. This corresponds to a time delay of 179 microseconds between the sounds arriving at the listening position. Figure 3 As shown, the spectrum at the listening position will show notches at odd multiples of f1 = 2.8 kHz.

[0044] Although this example consists of only two sound sources 202 and 204, the principle is the same for any number of sound sources greater than two. The notch pattern in the frequency response becomes more complex, with notches occurring at frequencies corresponding to the time delay of each secondary sound source, as well as odd harmonics of these frequencies.

[0045] When the primary and secondary sound sources are loudspeakers, the distance d1 between the acoustic centers of the sound sources may typically be between 15 cm and 30 cm. When the primary and secondary sound sources are drive units in a loudspeaker, the distance d1 between their acoustic centers is less than 5 cm. The farther the acoustic centers of the sound sources are apart, the lower the frequency at which the comb filtering extends, thus losing headroom in the input signal of the high-frequency shelf filter. However, the upper limit of the distance d1 between the acoustic centers of the sound sources depends on the listening distance d2; as the listening distance increases, the sound sources can be further separated.

[0046] To reduce the effects of comb filtering, the present invention applies a low pass filter to the secondary sound source 204 so that only the primary sound source 202 operates at a frequency where destructive interference will occur. However, this will result in a mismatch in the SPL at frequencies above and below the low pass (above and below f1) since one sound source is effectively above the low pass and both sound sources are below the low pass.

[0047] Fortunately, if Figure 4 As shown, above 1kHz, the energy in the music content generally decreases with frequency. Figure 4Different data sets from "Active speakers" by Stuart, JR (2006) in Proceedings of the 21st AES UK Conference: Home Audio are shown. Data set IEC268-1 is the IEC standard noise spectrum for power testing audio products, data set Sivian and Adams refers to previous research, and data set JRS is a data analysis performed by the author of this article. It is therefore clear that the reduction in energy above 1kHz is a common phenomenon in music content, as there is a general reduction in energy above 1kHz and energy below 100Hz in all four different data sets. This reduction in energy at higher frequencies provides potential processing headroom for compensating for the fact that there is only one source participating above the cutoff frequency of the low pass filter mentioned above. In order to achieve this compensation, a corresponding high frequency shelf filter is applied to the main sound source 202.

[0048] According to the rule g = 20log 10 (N+1), the gain of the high frequency shelf filter will depend on the number of secondary sources, where g is the gain of the shelf filter in decibels and N is the number of secondary sources. Figure 5 Possible responses of the low pass filter and high frequency shelf filter for secondary sources N=1 and N=2 are shown. Figure 5 The solid line in represents a possible response of a high frequency shelf filter for N=1, the dashed line represents a possible response of a high frequency shelf filter for N=2, and the dotted line represents a possible response of a low pass filter.

[0049] Figure 6 It is shown that a typical low pass filter 604 and high frequency shelf filter 602 will have characteristic transition frequencies that may be similar to, but not necessarily identical to, the first notch frequency f1 and will be at f1 or within a small frequency extension of f1. The characteristic frequencies of the low pass filter and high frequency shelf filter can be predicted by the previously calculated f1 608. Figure 6 As shown, typically, the characteristic frequency f of the high frequency shelf filter 606 is c1 will be slightly lower than f1 608, and the characteristic frequency f of the low pass filter 610 c2 It will be slightly higher than f1 608. However, the exact frequency will need to be tuned by one skilled in the art based on the specific system and implementation.

[0050] like Figure 4 As shown, the peak levels of frequencies in the music fall off rapidly above 1kHz, which provides headroom for applying a high frequency shelving filter, as most real systems are unlikely to exhibit destructive interference below 1kHz. However, care must be taken that the system has appropriate protection to prevent damage to the sound source in the event of atypical signals.

[0051] Therefore, the present invention is directed to a method of utilizing this headroom to reduce interference between multiple coherent sources while maintaining overall spectral balance.

[0052] Figure 7 This embodiment of three sound sources is shown: one primary sound source 710 and two secondary sound sources 712 and 714 . Figure 7 An audio signal 702 for the channels of an audio system is shown being split at 704 into a drive signal for a main sound source 710 and drive signals for two secondary sound sources 712 and 714. A high frequency shelf filter 706 is applied to the drive signal for the main sound source 710 and a low pass filter 708 is applied to the drive signals for the secondary sound sources 712 and 714.

[0053] like Figure 8 As shown, another embodiment introduces an all-pass filter 816 into the main sound source 810. Figure 8 An audio signal 802 for a channel of an audio system is shown to be split at 804 into a drive signal for a main sound source 810 and a drive signal for two secondary sound sources 812 and 814. A high frequency shelf filter 806 and an all-pass filter 816 are applied to the drive signal of the main sound source 810, and a low-pass filter 808 is applied to the drive signals of the secondary sound sources 812 and 814. The all-pass filter 816 newly introduced to the main sound source 810 is to compensate for the phase shift of the low-pass filter 808 on the secondary sound sources 812 and 814. For example, a second-order low-pass filter 808 results in a 180-degree phase shift near the center frequency of the filter. Therefore, a first-order all-pass filter 816 can be applied to the main sound source 810 to apply a complementary 180-degree phase shift. Therefore, the center frequency of the all-pass filter 816 should be similar to the center frequency for the low-pass filter 808.

[0054] The third preferred embodiment, as Fig. 9 As shown, additional all-pass filters 918 and 920 are introduced into the primary sound source 910 and the secondary sound sources 912 and 914. Fig. 9 An audio signal 902 for a channel of an audio system is shown to be split at 904 into a drive signal for a main sound source 910 and a drive signal for two auxiliary sound sources 912 and 914. A high frequency shelf filter 906, an all-pass filter 916, and an additional all-pass filter 918 are applied to the drive signal of the main sound source 910, and a low-pass filter 908 and an all-pass filter 920 are applied to the drive signals of the auxiliary sound sources 912 and 914. The newly introduced all-pass filters 918 and 920 can be used to improve the time alignment between the first and second drive signals and reduce the comb filter frequency cancellation effect. For example, the all-pass filter on the auxiliary sound source can be applied below the first notch frequency (f1), and the all-pass filter on the main sound source can be applied above the first notch frequency (f1) so as to reduce the cancellation at the first notch frequency by inverting the phase relationship.

[0055] Fig.10 1002 shows the frequency response when no filter is applied. The dashed line 1004 shows the frequency response when only a low pass filter and a high frequency shelf filter are applied (e.g., Figure 7 Dashed line 1006 shows the frequency response when an all-pass filter on the main source is applied in addition to the low-pass filter and high-frequency shelf filter (as shown in FIG. Figure 8 Solid line 1008 shows the frequency response when an additional all-pass filter is added to the primary and secondary sound sources in addition to all other filters applied (as shown in FIG. Fig. 9 As shown in Figure 3, all proposed filter combinations significantly reduce the spectral variation. However, when an additional all-pass filter is applied, it can be seen that the spectral variation is even further reduced compared to other combinations of filters.

[0056] In addition, if Fig.11A and 11B As shown, the proposed invention not only improves the frequency response at the listening position, but also reduces the spectral variation in space. Fig.11A Shown is the change in sound pressure level in the space when no filter is applied. Fig. 11B It shows that when the application Fig. 9 Variation of the sound pressure level in the space for all filters shown. Fig.11A and Fig. 11B The horizontal axis 1102 of both represents the off-axis distance of the listening position in the plane of the sound source array. The vertical axis 1104 represents the distance from the listening position to the array. The contour lines in the figure represent the SPL at that position in decibels, and each line represents a 3 decibel (dB) reduction in the SPL. Some contour lines representing multiples of 6dB reduction are labeled as such.

[0057] from Fig.11A It can be seen that when no filter is applied, as shown in the adjustment of the contour lines, there is significant destructive interference, with areas of high SPL marked as 1110. In contrast, Fig. 11B When the application is Fig. 9 When filtering is shown, there is no adjustment in the contours and the SPL drops uniformly.

[0058] In a preferred embodiment, the low pass, high frequency shelf and all pass filters are dual pole, dual zero digital second order filters, the design of which is well known to those skilled in the art. Such filters are preferred due to their simplicity of implementation, computational efficiency, and support in many existing signal processing systems. However, more complex designs for the filters may be used, and the filters may be implemented in software or hardware and in the analog or digital domain.

[0059] In some embodiments, the filter may be implemented as an update or enhancement to an existing system, or as part of a new system design. Additionally, in some embodiments, the filter will be implemented internally in the system, e.g. Figure 1 The filters are shown as being within each speaker, while in other embodiments the filters will be applied externally to the pre-processor device.

[0060] In order to maintain symmetry in the radiated sound field, an odd number of sound sources is preferred. In addition, in order to maximize the efficiency of the filter and limit the gain required for the shelf filter, the preferred number of sources is three. However, the present invention is applicable to any number of very close sound sources greater than one.

Claims

1. A method of generating a signal for driving a first line array of a plurality of coherent sound sources, wherein the first line array of a plurality of coherent sound sources comprises a primary sound source and one or more secondary sound sources, the method comprising the following steps: receiving an audio signal for a first channel of an audio system; deriving a first drive signal for driving the primary sound source and a second drive signal for driving the one or more secondary sound sources from the audio signal; applying a low pass filter to the second drive signal for driving the one or more secondary sound sources, the low pass filter reducing energy at a first frequency at which destructive interference would otherwise occur between sound from the primary sound source and the one or more secondary sound sources; as well as A corresponding high frequency shelf filter is applied to the first drive signal for driving the main sound source, the high frequency shelf filter compensating for the energy reduction at the first frequency caused by the low pass filter, wherein the gain g of the high frequency shelf filter is g=20log 10 (N+1), where N is the number of secondary sound sources.

2. The method of claim 1 further comprising applying a first all-pass filter to the first drive signal to compensate for additional interference introduced by the relative phase responses of the low-pass filter and the high-frequency shelf filter, resulting in energy loss around the characteristic frequency of the filter.

3. The method of claim 1 or 2, 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 to improve the time alignment between the first drive signal and the second drive signal.

4. The method of claim 1 , wherein a characteristic frequency of each of the low pass filter and the high frequency shelf filter is the inverse of two times a time delay between sounds arriving at a listening position from the primary sound source and the one or more secondary sound sources.

5. The method of claim 1, wherein the first line array of multiple coherent sound sources is a first line array of speakers comprising a primary speaker and one or more secondary speakers.

6. A computer program product comprising computer executable code which, when executed on one or more processors of an audio system, causes the system to perform the method of any one of claims 1 to 5.

7. The computer program product of claim 6, implemented as an update or enhancement to an existing digital signal processor sound source system.

8. The computer program product of claim 6, implemented as an update or enhancement to an existing multi-channel or stereo audio processor.

9. An audio system comprising one or more digital signal processors adapted to perform the method of any one of claims 1 to 5.

10. The audio system of claim 9, the high frequency shelf filter being implemented by a digital signal processor associated with the primary sound source, and the low pass filter being implemented by at least one digital signal processor associated with the one or more secondary sound sources.

11. An audio system for generating a flat sound field, the audio system comprising: A first line array of a plurality of coherent sound sources comprising a primary sound source and one or more secondary sound sources, wherein: 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; and deriving the first drive signal and the second drive signal from a received audio signal for a first channel of the audio system; applying a low pass filter to the second drive signal, the low pass filter reducing energy at a first frequency at which destructive interference would otherwise occur between sound from the primary sound source and the one or more secondary sound sources; and A corresponding high frequency shelf filter is applied to the first drive signal, the high frequency shelf filter compensating for the energy reduction at the first frequency caused by the low pass filter, wherein the gain g of the high frequency shelf filter is g=20log 10 (N+1), where N is the number of secondary sound sources.

12. The audio system of claim 11 further comprising a first all-pass filter applied to the first drive signal for compensating for additional interference introduced by the relative phase responses of the low-pass filter and the high frequency shelf filter, resulting in energy loss around a characteristic frequency of the filter.

13. The audio system of claim 12, further comprising additional, different second and third all-pass filters applied to the first drive signal and the second drive signal for improving time alignment between the first drive signal and the second drive signal.

14. The audio system of claim 11, wherein a characteristic frequency of each of the low pass filter and the high frequency shelf filter is the inverse of two times a time delay between sounds arriving at a listening position from the primary sound source and the one or more secondary sound sources.

15. The audio system of claim 11, wherein the first line array of a plurality of coherent sound sources is for mounting in a wall.

16. The audio system of claim 11, wherein the first line array of multiple coherent sound sources is arranged vertically or horizontally.

17. The audio system of claim 11 further comprising a second line array of a plurality of coherent sound sources driven by third and fourth drive signals, the third and fourth drive signals being derived from received audio signals for a second channel of the audio system in the same manner as the first and second drive signals and filtered in the same manner as corresponding signals in the first channel.

18. The audio system of claim 17, further comprising at least one additional line array of a plurality of coherent sound sources, the at least one additional line array comprising a primary sound source and one or more secondary sound sources driven by fifth and sixth drive signals, the fifth and sixth drive signals being derived from received audio signals for at least one additional channel of the audio system in the same manner as the first and second drive signals and filtered in the same manner as the corresponding signals in the first channel.

19. An audio system as claimed in any one of claims 11 to 18, wherein the first line array of a plurality of coherent sound sources is a first line array of a plurality of coherent loudspeakers comprising a primary loudspeaker and one or more secondary loudspeakers.

20. The audio system of claim 19, wherein The first line array of a plurality of coherent loudspeakers is arranged such that a distance between the acoustic centres of each subsequent loudspeaker in the first line array of a plurality of coherent loudspeakers is between 15 cm and 30 cm.

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