Patent ID: 12200438

DETAILED DESCRIPTION

The present invention may be implemented in a number of different ways according to the audio system being used. The following describes some example implementations with reference to the figures.

This invention is intended to alleviate the effect of spatial aliasing between two or more sound sources in close proximity. The invention is necessary when the source signals for each close proximity sound source are coherent, such as when using multiple loudspeakers as a single channel within a home theatre system100as shown inFIG.1.

Whilst in the example system inFIG.1the loudspeakers are mounted vertically; they could also be horizontally mounted. Furthermore, the centre set of loudspeakers104is not requisite, the system could be a stereophonic system consisting of only the left106and right108sets of loudspeakers, or indeed the system could be monophonic and consist of just one of the sets of loudspeakers. The right set of speakers108is made up of loudspeakers108a,108band108c. One of these will be a primary loudspeaker and two will be secondary loudspeakers. Additionally, whilst the sound sources in this example are two-way in-wall loudspeakers, the current invention could be applied to any close-proximity, coherent sound sources.

To demonstrate the problem this invention seeks to overcome, consider the system200given inFIG.2.FIG.2shows a simple example of two sound sources,202and204, with a distance d1metres between their acoustic centres. The listening position206, marked by an ‘X’, is d2metres from one of the sound sources, namely the primary sound source202, and is located both horizontally and vertically on-axis relative to this sound source. From Pythagoras' theorem, it is clear that the distance to the other sound source, the secondary sound source204, d3=√{square root over (d12+d22)} is larger than d2. Therefore, this gives rise to a time delay,

Δ⁢t=d3-d2c
seconds, where c=343 m/s is the speed of sound in air at 20 degrees Celsius, between the sounds arriving from the primary202and secondary204sound sources at the listening position206. This results in a series of notches in the frequency response observed at the listening position206due to destructive interference between the primary202and secondary204sources. This is known as “comb-filtering”. The notches will occur at frequencies

fn=n2⁢Δ⁢t
Hz, where n is all odd integers.

This comb-filtering effect is shown inFIG.3which plots frequency against sound pressure level relative to a single sound source. The notches of the “comb” shown inFIG.3are destructive interference occurring between the two sound sources202and204. The first notch302is at f1, the second notch304is at f3, the third notch306is at f5, and so on.

For example, a distance of 50 centimetres between the primary202and secondary204sound sources, with a listening position206that is 2 metres in front of the primary sound source202, results in a path length difference of 6.15 centimetres. This corresponds to a time delay between the sounds arriving at the listening position of 179 microseconds. Therefore, the frequency spectrum at the listening position will exhibit notches at odd multiples of f1=2.8 kHz, as shown inFIG.3.

Whilst this example only consists of two sound sources,202and204, the principle is the same for any number of sound sources greater than two. The pattern of notches in the frequency response simply gets more complex, with notches appearing at frequencies corresponding to the time delay to each secondary source, and odd harmonics of these frequencies.

When the primary and secondary sound sources are loudspeakers, the distance d1between the acoustic centres of the sound sources may typically be between 15 cm and 30 cm. When the primary and secondary sound sources are drive units within one loudspeaker, the distance d1between their acoustic centres may be as little as 5 cm. The further apart the acoustic centres of the sound sources are, the lower in frequency the comb filtering stretches and so headroom in the input signals for the high frequency shelving filter is lost. However, the upper limit of the distance d1between the acoustic centres of the sound sources depends on the listening distance d2; with larger listening distances the sound sources can be further apart.

To reduce the effect of the comb-filtering, the invention applies a low-pass filter to the secondary sound sources204so that only the primary sound source202is operating at frequencies where destructive interference will occur. However, this will lead to a mismatch in the SPL at frequencies above and below the low-pass (above and below f1) due to effectively having one sound source above the low-pass and two below it.

Fortunately, there is a general reduction with frequency in energy in music content above 1 kHz, as shown inFIG.4, which presents different data sets from Stuart, J. R. (2006). “Active loudspeakers”, In Proceedings of the 21st AES UK Conference: Audio at Home. The data set IEC268-1 is an IEC standard noise spectrum for power testing audio products, the data sets Sivian and Adams relates to previous studies and the data set JRS is data analysis carried out by the author of the paper. It is therefore clear that this reduction in energy above 1 kHz is a common occurrence in music content as in all four different data sets there is a general reduction in energy above 1 kHz and energy below 100 Hz. This reduction in energy at higher frequencies offers potential processing headroom for compensating for the fact that above the aforementioned low-pass filter cut-off frequency there is only one source contributing. To implement this compensation, a corresponding high frequency shelving filter is applied to the primary sound source202.

The gain of the high-frequency shelving filter will depend on the number of secondary sources according to the rule g=20 log10(N+1), where g is the gain of the shelving filter in decibels and N is the number of secondary sources.FIG.5shows possible responses for the low-pass filter and high frequency shelving filter for N=1 and N=2 secondary sources. The solid line inFIG.5shows a possible response for the high frequency shelving filter for N=1, the dashed line shows a possible response for the high frequency shelving filter for N=2, and the dotted line shows a possible response for the low-pass filter.

FIG.6shows that typically both the low-pass filter604and the high-frequency shelving filter602will have a characteristic transition frequency which may be similar, but not necessarily the same as f1and will be at or within a small frequency spread of f1, the first notch frequency. The characteristic frequencies of both the low-pass filter(s) and the high-frequency shelving filter can be predicted by the previously calculated f1608. As shown inFIG.6, typically the characteristic frequency of the high frequency shelving filter606, fc1, will lie slightly below f1608and the characteristic frequency of the low-pass filter(s)610, fc2, will lie slightly above f1608. However, the exact frequencies will require tuning by one skilled in the art, based on the specific system and implementation.

As demonstrated inFIG.4, the peak level of frequencies in music rapidly drops off above 1 kHz, which affords headroom for applying the high frequency shelving filter, as most real-world systems are unlikely to exhibit destructive interference below 1 kHz. Nevertheless, careful attention must be taken that the system has appropriate protection to prevent damage to the sound sources in case of atypical signals.

Therefore, the present invention relates to methods taking advantage of this headroom in order to reduce interference between multiple coherent sources, whilst maintaining overall spectral balance.

FIG.7illustrates such an embodiment for three sound sources: one primary710and two secondary,712and714.FIG.7shows that an audio signal702for a channel of an audio system is split704into a drive signal for a primary sound source710and a drive signal for two secondary sound sources,712and714. A high-frequency shelving filter706is applied to the drive signal for the primary sound source710and a low-pass filter708is applied to the drive signal for the secondary sound source,712and714.

A further embodiment, shown inFIG.8, introduces an all-pass filter816to the primary sound source810.FIG.8shows that an audio signal802for a channel of an audio system is split804into a drive signal for a primary sound source810and a drive signal for two secondary sound sources,812and814. A high-frequency shelving filter806and an all-pass filter816are applied to the drive signal for the primary sound source810and a low-pass filter808is applied to the drive signal for the secondary sound sources,812and814. The newly introduced all-pass filter816to the primary sound source810is in order to compensate for the phase-shift of the low-pass filter808on the secondary sound source,812and814. For example, a second order low-pass filter808results in a 180 degree phase-shift about the centre frequency of the filter. A first order all-pass filter816could therefore be applied to the primary sound source810, in order to apply a complementary 180 degree phase-shift. As such, the centre frequency of the all-pass filter816should be similar to that used for the low-pass filter808.

A third, and preferred embodiment, as shown inFIG.9, introduces additional all-pass filters,918and920, to both the primary910and secondary,912and914, sound sources.FIG.9shows that an audio signal902for a channel of an audio system is split904into a drive signal for a primary sound source910and a drive signal for two secondary sound sources,912and914. A high-frequency shelving filter906, an all-pass filter916and an additional all-pass filter918are applied to the drive signal for the primary sound source910and a low-pass filter908and an all-pass filter920are applied to the drive signal for the secondary sound sources,912and914. The newly introduced all-pass filters,918and920, can be used improve the time-alignment between the first and second drive signals, reducing the comb-filter frequency cancellation effect. For example, the all-pass filter on the secondary sound source can be applied below the frequency of the first notch (f1), while the all-pass filter on the primary sound source can be applied above the frequency of the first notch (f1), in order to reduce the cancellation at the first notch frequency by inverting the phase relationship.

FIG.10shows a simulated frequency response at the listening position without the filters proposed by this invention and with the different combinations of filters suggested above. The dotted line1002shows the frequency response when no filters are applied. The dashed-dotted line1004shows the frequency response when just the low-pass filter and the high frequency shelving filter are applied (as inFIG.7). The dashed line1006shows the frequency response when the all-pass filter on the primary source is applied in addition to the low-pass filter and the high frequency shelving filter (as inFIG.8). The solid line1008shows the frequency response when the additional all-pass filters are added to both the primary and secondary sound sources, in addition to all other filters is applied (as inFIG.9). It can be seen that all combinations of filters proposed significantly reduce the spectral variation. However, when the further all-pass filters are applied, it can be seen that the spectral variation is even further reduced compared to the other combinations of filters.

Additionally, as shown inFIGS.11A and11B, the proposed invention not only improves the frequency response at the listening position, but also reduces spectral variation across space.FIG.11Ashows the variation in the sound pressure level across space when no filters are applied.FIG.11Bshows the variation in the sound pressure level across space when all the filters, as set out inFIG.9, are applied. The horizontal-axis1102of bothFIG.11AandFIG.11Brepresents the distance off-axis of the listening position in the plane of the sound source array. The vertical-axis1104represents the distance of the listening position away from the array. The contour lines within the plots represents the SPL at that positon in decibels, with each line representing a decrease in SPL of 3 decibel (dB). Some contours representing a multiple of 6 dB decrease have been labelled as such.

As can be seen fromFIG.11A, when no filters are applied there is significant destructive interference, as illustrated in the modulation of the contour lines, with regions of high SPL labelled as1110. Conversely, inFIG.11B, when there is filtering applied, as set out inFIG.9, there is an absence of modulating in the contour lines and the SPL falls off uniformly.

In the preferred embodiment the low-pass, high-frequency shelving and all-pass filters are two-pole, two-zero digital biquad filters, the design of which is known to someone skilled in the art. Such filters are preferred due to the fact that the implementation of these filters is simple, computationally efficient and supported on many existing signal processing systems. However, more complex designs for the filters could be used and the filters can be implemented in software or hardware as well as in the analogue or digital domains.

In some embodiments the filters may be implemented as an update or enhancement to an existing system, or as part of the design of a new system. Additionally, in some embodiments the filters will be implemented internally to the system, for example within each of the loudspeakers shown inFIG.1, whereas in other embodiments the filters will be applied externally in a pre-processor device.

Odd numbers of sound sources are preferred, in order to maintain symmetry in the radiated sound field. Furthermore, the preferred number of sources is three in order to maximise the effectiveness of the filters and limit the required gain of the shelving filter. However the current invention could be applied to any number of close proximity sound sources greater than one.