This invention relates to a directional sound system and a method for processing an input signal to a directional sound system.
The ability to control sound radiation patterns in entertainment, gaming, communication and personal messaging is becoming an important differentiating feature in many commercial products. A common aim in these systems is to create a highly-directional sound field to targeted audiences by forming a tune-in zone (or personal audio) for a group of people. There are several ways to generate the directional sound field. These include (i) using a sound dome that projects sound to a convex surface to focus sound waves to the listeners below the sound dome; (ii) using a loudspeaker array with the phase-amplitude differences between different loudspeakers adjusted to spatially steer an audible sound beam in a horizontal plane; and (iii) modulating an audible sound signal onto an ultrasonic carrier signal and projecting the modulated signal via special types of ultrasonic emitters to generate a parametric array through the air in such a way that audible sound can travel in a column of sound beam. Loudspeakers generating the directional sound field using (iii) are commonly called parametric (or ultrasonic) loudspeakers. The parametric loudspeaker is based on a nonlinear acoustics property (known as the parametric array effect in air) that uses ultrasound signal to carry the audible sound signal in a tight beam, just like an audio spotlight.
When using a loudspeaker array (as described in (ii)) to steer an audible sound beam at low frequencies, for example, at frequencies less than 200 Hz, the dimension of the loudspeaker array must be significantly greater than the audio wavelength in order to achieve a good directivity. Usually, this means that the dimension of the loudspeaker array must be more than a meter in diameter. This approach of creating a focused sound beam hence incurs a high cost since a large loudspeaker array is required. In contrast, a parametric loudspeaker (as described in (iii)) is able to generate a highly-directional sound beam for a low-frequency sound wave whose wavelength is much larger than the loudspeaker diameter. This is because the small-sized ultrasonic emitter in the parametric loudspeaker is able to produce a highly-directional sound beam without using a vibrating cone as opposed to conventional loudspeakers.
The Berktay far-field model is widely used to approximate the nonlinear sound propagation by the parametric loudspeaker through the transmission medium. This model uses an expression as shown in Equation (1) to predict the far field array response of the parametric loudspeaker. According to Equation (1), the demodulated signal (or audible difference frequency) pressure p2(t) along the axis of propagation is proportional to the second time-derivative of the square of the envelope of the modulated signal when amplitude modulation is used. In Equation (1), β is the coefficient of nonlinearity, P0 is the primary wave pressure, a is the radius of the ultrasonic emitter, ρo is the density of the transmission medium, c0 is the small signal sound speed, z is the axial distance from the ultrasonic emitter, α0 is the attenuation coefficient at the source frequency and E(t) is the envelope of the modulated signal.
As shown in Equation (1), the nonlinear sound propagation results in a distortion in the demodulated signal. This in turn results in a distortion in the audible signal generated by the parametric loudspeaker, hence affecting the performance of the parametric loudspeaker. Furthermore, the current parametric loudspeaker technology is severely limited by the technological constraints of ultrasonic emitters. One such technological constraint is the small usable low-frequency bandwidth of the ultrasonic emitters.
Digital signal processing techniques have previously been proposed to overcome the technological limitations of the parametric loudspeaker technology. These techniques usually involve pre-processing algorithms which can be programmed in a digital signal processor to enhance, equalize and compensate for any distortion in the audio quality of the signal before sending the processed signal to the ultrasonic emitter. Examples of such techniques are described below.
As shown in
a′m(t)=−2(√{square root over (E1(t))}−E2(t))×(t−m) (2)
am(t+1)=am(t)+βa′m(t) (3)
The output 40 of the adaptive filter 208 is shown in Equation (4) as follows.
A pre-processing technique is also proposed in U.S. Pat. No. 6,584,205 (hereinafter, Croft) to improve the performance of parametric loudspeakers.
According to an exemplary aspect, there is provided a directional sound system comprising: a plurality of equalization stages configured to equalize an input signal; and a transducer stage configured to transmit the equalized input signal; wherein the plurality of equalization stages comprises a first equalization stage configured to employ an approximated model of the transducer stage and a second equalization stage configured to compensate for differences between the approximated model of the transducer stage and an actual model of the transducer stage.
According to another exemplary aspect, there is provided a method for processing an input signal to a directional sound system, the method comprising: repeatedly equalizing the input signal; and transmitting the equalized input signal; wherein a first equalization of the input signal is performed using an approximated model of the transmission and a second equalization of the input signal comprises compensating for the differences between the approximated model of the transmission and an actual model of the transmission.
Having more than one equalization stage is advantageous as the first equalization stage can provide a coarse equalization of the input signal whereas the second equalization stage can provide a finer equalization of the input signal. In this way, the equalization of the input signal may be performed in a more efficient and accurate manner.
Preferably, the directional sound system further comprises a modulation stage configured to modulate the equalized input signal from the first equalization stage prior to the second equalization stage, wherein the modulation stage employs a modulation technique which uses a pre-distortion term with a variable order. Similarly, the method preferably further comprises modulating the equalized input signal from the first equalization prior to the second equalization by employing a modulation technique which uses a pre-distortion term with a variable order.
It is advantageous to employ the modulation technique which uses a pre-distortion term with a variable order. The addition of the pre-distortion term may reduce distortion in the demodulated signal (i.e. the audio signal output of the directional sound system). The amount of reduction in the distortion is dependent on the order of the pre-distortion term. A higher order will achieve a greater amount of reduction in the distortion. However, a higher order pre-distortion term requires an ultrasonic transducer with a higher bandwidth. By using a pre-distortion term with a variable order, the flexibility of the modulation technique is increased and the order of the pre-distortion term may be varied to suit the requirements of the ultrasonic transducer used in the directional sound system. For example, a lower order may be used for ultrasonic transducers with lower bandwidth whereas the order may be scaled up for ultrasonic transducers with higher bandwidth to further reduce the distortion in the audio signal output of the directional sound system.
Preferably a sub-band approach is employed whereby the input signal is split into a plurality of frequency regions and each frequency region of the input signal is processed independently through at least one stage of the directional sound system.
Using the sub-band approach, the linear nature of the frequency and phase response of the transducer stage within each sub-band may be exploited during equalization. Furthermore, since equalization may be applied to each frequency region independently, the amplitude of the equalized signal in each frequency region will generally not be as low as the amplitude of the equalized signal in the full-band approach and thus, a lower amplification is required for the equalized signal in each frequency region. Furthermore, by using the sub-band approach, the input signal may be downsampled, thus lowering and varying the speed requirement for processing each frequency region and in turn lowering the speed requirement for processing the entire signal. This mixed-rate processing technique thus removes the need for high-end processors and instead, a low cost digital signal processor can be used to implement the directional sound system.
By employing both the sub-band approach and the modulation technique using the pre-distortion term with a variable order, the modulation technique for each frequency region may be adjusted independently. Thus, different components with different requirements (for example, different modulation techniques or different ultrasonic transducers) may be used for different frequency regions and the modulation technique for each frequency region may be adjusted to match the requirements of the components used for that frequency region. Thus, the combination of the flexible modulation technique and the sub-band approach in the embodiments of the present invention is extremely advantageous.
Preferably, the modulation stage comprises a first modulation stage and the directional sound system further comprises: a second modulation stage configured to further modulate the modulated equalized input signal wherein a carrier frequency of the further modulated equalized input signal is dependent on a first carrier frequency in the first modulation stage and a second carrier frequency in the second modulation stage.
Having more than one modulation stage is advantageous as it divides the complexity of the system into more parts and hence, the system can be realized with relatively cheaper hardware such as analog modulators. Furthermore, using more modulation stages offers more flexibility in the selection of the carrier frequency for the modulation of the input signal since the overall carrier frequency may be adjusted by independently adjusting the carrier frequency of each modulation stage.
In order that the invention may be fully understood and readily put into practical effect there shall now be described by way of non-limitative example only exemplary embodiments, the description being with reference to the accompanying illustrative drawings.
In the drawings:
The input signal 414 of the parametric loudspeaker system 400 is usually an audible sound signal. As shown in
To achieve an optimal reduction of distortion in the audible signal produced by the parametric loudspeaker system 400, it is preferable that both the frequency and phase response of the transducer stage 412 are compensated in the equalization stages 404, 406 of the parametric loudspeaker system 400.
The parametric loudspeaker system 400 employs a sub-band approach whereby the input signal is split into a plurality of frequency regions (in other words, a plurality of bands) by the filter bank 402 and each frequency region of the input signal is independently processed through the first equalization stage 404, and the modulation stage 408. Thus, the parametric loudspeaker system 400 may be referred as a “multi-band audio beaming” system.
The different stages of the parametric loudspeaker system 400 will now be described in more detail.
The filter bank 402 serves to split the input signal 414 into different frequency regions. As shown in
The first equalization stage 404 serves to compensate for one or more expected changes in the input signal after demodulation. In one example, the first equalization stage 404 serves to compensate for an expected 12 dB/octave slope change in the input signal after demodulation as predicted by the Berktay's approximation in Equation (1). This change arises due to the second time-derivative in Equation (1). The first equalization stage 404 further serves to compensate for the frequency and phase response of the transducer stage 412 which is usually highly non-linear. As shown in
The modulation stage 408 employs a modulation technique which uses a pre-distortion term with a variable order as shown in
As shown in
with a second carrier signal cos ω0t to produce a compensation signal, and summing the main signal and the compensation signal to generate the output ĝ(t). Note that the first and second carrier signals are orthogonal to each other and that the pre-distortion term is generated by the signal generator 502 whereby the order of the signal generator 502 represents the order of the pre-distortion term it generates. From Equation (5), it can be seen that as compared to a typical DSBAM scheme which merely generates the main signal (1+mg(t))sin ω0t, the output ĝ(t) comprises an additional orthogonal term
The addition of the pre-distortion term can reduce the distortion in the demodulated signal. This is elaborated below. Denoting f1(t)=1+mg(t) and the output of the signal generator 502 as f2(t), the output ĝ(t) of the band n modulator can be written in the form as shown in Equation (6).
In other words, the envelope of the modulation technique output ĝ(t) is √{square root over (f12(t)+f22(t))}{square root over (f12(t)+f22(t))}. According to the Berktay's approximation (Equation (1)), the demodulated signal (or audible difference frequency) pressure p2(t) along the axis of propagation is proportional to the second time-derivative of the square of the envelope of the modulated signal. Substituting √{square root over (f12(t)+f22(t))}{square root over (f12(t)+f22(t))} into Equation (1), Equation (7) is obtained as follows.
As shown in Equation (8), by setting f2(t)=√{square root over (1−m2g2(t))}, the demodulated signal becomes proportional to the input signal g(t). In other words, the distortion in the demodulated signal is completely removed. However, this is only true if and only if the ultrasonic transducer 412 has infinite bandwidth. As this is not the case with practical ultrasonic transducers, the pre-distortion term f2(t)=√{square root over (1−m2g2(t))} is approximated using its truncated Taylor series
By adjusting the value of q, the order of the pre-distortion term
can be varied.
In the parametric loudspeaker system 400, the modulation stage 408 comprises a plurality of band n modulators with each band n modulator employing the modulation technique of
The frequency spectrum of the ultrasonic transducer in the transducer stage 412 is generally non-symmetrical about its resonance frequency. The second equalization stage 406 serves to compensate for this. The second equalization stage 406 further serves to compensate for the differences between an actual model of the transducer stage 412 and the approximated model of the transducer stage 412 used in the first equalization stage 404. The actual model of the transducer stage 412 may be obtained through experimentation.
As shown in
As shown in
The parametric loudspeaker system 600 extends the sub-band approach to the second equalization stage 602, the amplification stage 604 and the transducer stage 606. In the parametric loudspeaker system 600, the input signal 608 is first split into different frequency regions using the filter bank 402′ and each frequency region of the input signal 608 is independently processed through the two equalization stages 404′, 602, the modulation stage 408′, the amplification stage 604 and the transducer stage 606. Such a structure allows the usage of different types of emitters in different ultrasonic transducers for different frequency regions. Different types of emitters may have different bandwidth or amplification requirements and the output of each band n modulator of the modulation stage 408′ may be independently adjusted to match the requirements of the respective ultrasonic transducer (Ultrasonic Transducer Group n) operably connected to it. The output of each band n modulator may also be independently adjusted according to the bandwidth of the filter hn in the filter bank 402′ operably connected to it.
For example, each band n modulator may employ the modulation technique as shown in
As shown in
Having more than one modulation stage is advantageous. In the case where there is only a single modulation stage, it is necessary to sample the input signal at a high sampling frequency if the input signal is to be modulated with a high carrier frequency. By providing an additional modulation stage, the carrier frequency in each modulation stage can be lowered without lowering the overall carrier frequency. Thus, the sampling frequency of the input signal may be lowered and the computational requirement for processing the input signal may be reduced. Furthermore, having two modulation stages 408″, 702 divides the complexity of the system 700 into two parts and hence, the system 700 can be realized with relatively cheaper hardware such as analog modulators. For example, users may implement the first modulation stage 408″ on a readily available computer despite its low sampling frequency since the overall carrier frequency may be increased using the second modulation stage 702 which may be implemented on, for example, inexpensive analog modulators external to the computer.
Furthermore, using the additional modulation stage 702 offers more flexibility in the selection of the carrier frequency for the modulation of the input signal since the overall carrier frequency may be adjusted by adjusting either the first carrier frequency in the first modulation stage 408″ or the second carrier frequency in the second modulation stage 702, or both of these carrier frequencies. This is particularly useful in the sub-band approach especially when the ultrasonic transducers used for different frequency regions have different resonance frequencies.
Other variations of the parametric loudspeaker system 400 may also be possible.
For example, a full-band approach may be adopted for any of the above embodiments whereby the input signal is processed as a whole through all the stages of the parametric loudspeaker system. In one example, the parametric loudspeaker system comprises a single modulator in the modulation stage and a single ultrasonic transducer in the transducer stage. The single modulator may employ the modulation technique as shown in
Alternatively, the parametric loudspeaker system may comprise a plurality of adaptive filters in the second equalization stage and only a single ultrasonic transducer in the transducer stage. In another example, the parametric loudspeaker system may comprise a single adaptive filter in the second equalization stage and a plurality of ultrasonic transducers in the transducer stage. However, this example is not preferable.
Furthermore, the filter bank and first equalization stage of the parametric loudspeaker system 400, 600, 700 may be combined into a single equalization stage whereby the single equalization stage serves to split the input signal into different frequency regions, compensate for one or more expected changes in the input signal after demodulation and at the same time, compensate for the frequency and phase response of the transducer stage. In addition, the parametric system may comprise more than two equalization stages and may also comprise more than two modulation stages. Each of these modulation stages may or may not employ the modulation technique of
The advantages of the embodiments of the present invention are as follows.
Preprocessing methods to reduce the distortion in parametric loudspeakers have previously been suggested. However, these preprocessing methods are based on a single-band approach, whereby a single pre-processing method and modulation technique is applied to the entire frequency range of the signal. Also, there is hardly any mention on the types of ultrasonic emitters used with these preprocessing methods. Through experiment, the inventors of this application found that different ultrasonic emitters have very different frequency responses that need to be individually addressed in order to best reproduce directional sound with minimum distortion. The embodiments of the present invention can address both the single-band problem and the problem arising due to the difference in the frequency responses of different ultrasonic emitters. An adaptive approach is also incorporated in the embodiments to compensate for the deficiency in the ultrasonic emitters.
In the embodiments of the present invention, the modulation stage employs a modulation technique known as Modified Amplitude Modulation q (MAMq) which uses a pre-distortion term with a variable order. In this modulation technique, an orthogonal term (formed by multiplying a pre-distortion term with an orthogonal carrier signal) is added to the usual DSBAM scheme. This is different from the typical amplitude-based modulation techniques used in the prior art.
Thus, the addition of the pre-distortion term can greatly reduce distortion in the demodulated signal (i.e. the audio signal output of the parametric loudspeaker system). As shown in
Furthermore, the embodiments of the present invention use a sub-band approach. Unlike the conventional fullband approach, the embodiments of the present invention are able to solve the problems in a more detailed manner by partitioning the input signal into smaller frequency regions (or bands) and using a “divide-and-conquer” approach to reduce the distortion found in these smaller regions. As such, different algorithms may be used to remove the distortion found in different frequency regions, thereby enhancing the quality of the audio sound produced by the parametric loudspeaker system.
Using the sub-band approach, the embodiments of the present invention can exploit the linear nature of the frequency and phase response of the transducer stage within each subband. Thus, as compared to the non-subband (i.e. fullband) approach, the compensation for the frequency and phase response of the transducer stage is simplified using the embodiments of the present invention.
The amplitude of the input signal may be higher in certain frequency regions whereas it may be lower in other frequency regions. Typically, equalization is achieved by lowering the amplitude of the input signal in high amplitude frequency regions to match the amplitude of the input signal in the lowest amplitude frequency region (i.e. the frequency region in which the amplitude of the input signal is the lowest). In order to compensate for the reduction in the signal level, the signal is amplified after equalization. This is highly undesirable due to the low efficiency in electrical to acoustic conversion in ultrasonic transducers. The sub-band approach in the embodiments of the present invention avoids these issues arising in typical full-band equalization. Using the sub-band approach, the equalization is applied to each frequency region independently and therefore the amplitude of the equalized signal in each frequency region will generally not be as low as the amplitude of the equalized signal in the full-band approach. Thus, a lower amplification is required for the equalized signal in each frequency region.
It can be shown that the embodiments of the present invention using the sub-band approach provide a significant reduction in harmonic distortion and in the intermodulation distortion as compared to the traditional full-band approach. Furthermore, by using the sub-band approach, the input signal may be downsampled, thus lowering and varying the speed requirement for processing each frequency region and in turn lowering the speed requirement for processing the entire signal. This mixed-rate processing technique thus removes the need for high-end processors and instead, a low cost digital signal processor can be used to implement the multi-band Audio Beaming System in the embodiments of the present invention. Also, although ultrasonic transducers with higher bandwidth are more desirable, they are usually more expensive. The sub-band approach allows the use of different types of ultrasonic transducers in the same system, thus allowing the use of cheaper ultrasonic transducers with lower bandwidth for input frequencies which are less important. This in turn lowers the cost of the system.
Furthermore, there is flexibility in the implementation of the embodiments in the present invention. For example, different variations of the parametric loudspeaker system 400 are possible. The embodiments of the present invention can also be scaled, for example by the manufacturer, to fit the required applications. The pricing of the system may also vary according to the scaling. Thus, the products can be differentiated.
In addition, in the embodiments of the present invention, two or more modulation stages may be provided. This allows the embodiments to be realized with relatively cheaper hardware such as analog modulators. This also offers more flexibility in the selection of the carrier frequency for the modulation of the input signal since the overall carrier frequency may be adjusted by independently adjusting the carrier frequency in each modulation stage.
Also, two or more equalization stages may be provided in the embodiments of the present invention. This is advantageous as the initial equalization stages can implement a coarser equalization of the input signal whereas the later equalization stages can implement a finer equalization of the input signal. In this way, equalization of the input signal is performed in a more efficient and accurate manner.
The embodiments of the present invention thus provide a comprehensive approach in reducing distortion in a parametric array. There are several commercial applications that can be achieved using the embodiments of the present invention. Some of these applications include (a) delivering private messages in a museum, billboard, art gallery, restaurant etc., (b) multi-lingual teleconferencing and messaging, (c) creating new binaural and three-dimensional effects in gaming and home entertainment, (d) directional loud hailer and (e) personal tune-in zone.
In summary,
Whilst the foregoing description has described exemplary embodiments, it will be understood by those skilled in the technology concerned that many variations in details of design, construction and/or operation may be made without departing from the present invention.
The present invention claims priority under 35 U.S.C. §§365 and 119 of Patent Cooperation Treaty International Application No. PCT/SG2010/000312 filed Aug. 25, 2010 and under 35 U.S.C. §119 to U.S. Provisional Patent Application Ser. No. 61/236,687 filed Aug. 25, 2009.
Filing Document | Filing Date | Country | Kind | 371c Date |
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PCT/SG2010/000312 | 8/25/2010 | WO | 00 | 2/17/2012 |
Publishing Document | Publishing Date | Country | Kind |
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WO2011/025461 | 3/3/2011 | WO | A |
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Number | Date | Country | |
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20120148053 A1 | Jun 2012 | US |
Number | Date | Country | |
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61236687 | Aug 2009 | US |