The present invention relates to a surround sound system for reproducing a spatial sound field within a room.
In home theatre, typical surround sound is performed using 5 or 7 loudspeakers plus a subwoofer, such as in the Dolby surround format. Such surround sound systems are able to create direct fields from various directions and ambient (diffuse) fields, but they cannot perform a full ambisonics reproduction that is required to recreate a sound over a spatial area or volume.
The more high-end and complex ambisonics surround sound systems typically require a large circular or spherical arrangement of loudspeaker drivers surrounding the sound control region to reproduce a spatial sound field. However, the requirement for such large arrays of loudspeakers is not compatible with the demands for compact surround sound systems in home theatre and entertainment systems.
A fundamental challenge to sound field control is the presence of room reverberation. Many current surround sound systems simply ignore the presence of room reverberation, although there are some possibilities for avoiding reverberation or cancelling reverberation outside the sound control region [4-8,22]
In this specification where reference has been made to patent specifications, other external documents, or other sources of information, this is generally for the purpose of providing a context for discussing the features of the invention. Unless specifically stated otherwise, reference to such external documents is not to be construed as an admission that such documents, or such sources of information, in any jurisdiction, are prior art, or form part of the common general knowledge in the art.
It is an object of the present invention to provide an improved compact surround sound system that is capable of reproducing spatial sound fields with a reduced number loudspeakers, or to at least provide the public with a useful choice.
In a first aspect, the present invention broadly consists in a surround sound system for reproducing a spatial sound field in a sound control region within a room having at least one sound reflective surface, comprising: multiple steerable loudspeakers located about the sound control region, each loudspeaker having a plurality of different individual directional response channels being controlled by respective speaker input signals to generate sound waves emanating from the loudspeaker with a desired overall directional response created by a combination of the individual directional responses; and a control unit connected to each of the loudspeakers and which receives input spatial audio signals representing the spatial sound field for reproduction in the sound control region, the control unit having pre-configured filters for filtering the input spatial audio signals to generate the speaker input signals for all the loudspeakers to generate sound waves with co-ordinated overall directional responses that combine together at the sound control region in the form of either direct sound or reflected sound from the reflective surface(s) of the room to reproduce the spatial sound field, the filters being pre-configured based on acoustic transfer function data representing the acoustic transfer functions measured in the sound control region from the individual directional responses of each of the loudspeakers at their respective locations in the room.
Preferably, the input spatial audio signals may be in an ambisonics-encoded surround format that is received and directly filtered by the filters in the control unit to generate the speaker input signals for the loudspeakers. Alternatively, the input spatial audio signals may be in a non-ambisonics surround format and the control unit further comprises a converter that is configured to convert the non-ambisonics input signals into an ambisonics surround format for subsequent filtering by the filters in the control unit to generate the speaker input signals for the loudspeakers.
Preferably, the control unit may be switchable between a configuration mode in which the control unit configures the filters for the room and a playback mode in which the control unit processes the input spatial audio signals for reproduction of the spatial sound field using the loudspeakers.
Preferably, the control unit may comprise a configuration module that is arranged to automatically configure the filters in the configuration mode based on input acoustic transfer function data for the room that is measured by a sound field recording system.
Preferably, the input acoustic transfer function data for the room may be measured by a sound field recording system comprising a microphone array located in the sound control region and the acoustic transfer function data represents the acoustic transfer functions measured by the microphone array in response to test signals generated by each of the loudspeakers for each of their directional responses. More preferably, the configuration module may receive raw measured acoustic transfer function data from the sound field recording system and converts it into an ambisonics representation of the acoustic transfer function data which is used to configure the filters of the control unit.
Preferably, the filters of the control unit may be ambisonics loudspeaker filters.
In one form, the surround sound system may be configured to provide a 2-D spatial sound field reproduction in a 2-D sound control region. Preferably, the sound control region may be circular and has a predetermined diameter. More preferably, the sound control region may be located in a horizontal plane and the loudspeakers are at least partially co-planar with the sound control region.
Preferably, each loudspeaker may be located within a respective loudspeaker location region, the room being radially and equally segmented into loudspeaker location regions about the origin of the sound control region based on the number of loudspeakers, and wherein each loudspeaker region is defined to extend between a pair of radii boundary lines extending outwardly from the origin of the sound control region. Preferably, the angular distance between each pair of radii boundary lines may correspond to 360°/L, where L is the number of loudspeakers.
Preferably, each loudspeaker may be spaced apart from every other loudspeaker by at least half of a wavelength of the lowest frequency of the operating frequency range of the surround sound system. This condition will ensure de-correlated room excitations above the Schroeder frequency.
Preferably, each loudspeaker may be spaced apart from any reflective surface(s) in the room by at least quarter of a wavelength of the lowest frequency of the operating frequency range of the surround sound system.
Preferably, each loudspeaker may be spaced at least 0.5 m from the perimeter of the sound control region. More preferably, each loudspeaker may be spaced at least 1 m from the perimeter of the sound control region.
Preferably, each loudspeaker may be configured to generate overall directional responses having up to Mth order directivity patterns, where M is at least 1. More preferably, each loudspeaker may be configured to generate overall directional responses having up to Mth order directivity patterns, wherein M is equal to 4. Typically, the value 2M+1 corresponds to the number of individual directional response channels available for each loudspeaker.
Preferably, each loudspeaker comprises at least an individual directional response channel corresponding to a first order directional response.
In one form, each loudspeaker may comprise at least individual directional response channels corresponding to 2M+1 phase mode directional responses.
In a preferred form, each loudspeaker may comprise at least individual directional response channels corresponding to an omni-directional response, and cos(mφ) and sin(mφ) for m=1, 2, . . . , M, and where φ is equal to the desired angular direction of the loudspeaker overall directional response relative to the origin of the loudspeaker.
Preferably, the overall directional response of each loudspeaker may be steerable in 360° relative to the origin of the loudspeaker.
Preferably, each loudspeaker may comprise multiple drivers configured in a geometric arrangement within a single housing, each driver being driven by a driver signal to generate sound waves, and wherein each loudspeaker further comprises a beamformer module that may be configured to receive and process the speaker input signals corresponding to the individual directional response channels of the loudspeaker and which generates driver signals for driving the loudspeaker drivers to create an overall sound wave having the desired overall directional response.
Preferably, each loudspeaker may comprise a housing within which a uniform circular array of monopole drivers of a predetermined radius are mounted, and wherein the number of drivers and radius may be selected based on the desired maximum order of directivity pattern required for the loudspeaker. More preferably, the monopole drivers may be spaced apart from each other by no more than half a wavelength of the maximum frequency of the operating frequency range of the surround sound system.
Preferably, the surround sound system may comprise at least four steerable loudspeakers.
Preferably, the control unit may be configured to automatically step-up the order of the directivity patterns of the overall directional responses of the loudspeakers as the frequency of the spatial sound field represented by input spatial audio signals increases to thereby maintain a substantially constant size of sound control region.
Preferably, the control unit may be configured to automatically step-up the order of the directivity pattern of the overall directional responses of the loudspeakers at predetermined frequency thresholds in the operating frequency range of the surround sound system, the thresholds being determined based on the number of loudspeakers and the desired size of sound control region.
Preferably, the loudspeakers may be equi-spaced relative to each other about the sound control region. More preferably, the loudspeakers may be sparsely located about the sound control region. Preferably, each loudspeaker may be located near a reflective surface, such as a wall in the room or in the vicinity of a corner of the room.
Preferably, the spatial sound field may be represented in the sound control region by direct sound in combination with first order, second order, and/or higher order reflections from sound waves reflected off one or more reflective surfaces of the room.
Preferably, the surround sound system may be configurable to reproduce higher order ambisonics spatial sound fields.
Preferably, the diameter of the sound control region may be at least 0.175 m. Typically, the diameter of the sound control region may be in the range of about 0.175 m to about 1 m.
In another form, the surround sound system may be configured to provide a 3-D spatial sound field reproduction in a 3-D sound control region. More preferably, the 3-D sound control region may be spherical in shape.
It will be appreciated that other shapes of 2-D and 3-D sound control regions could alternatively be used, but typically using a sound control region that is a circular (spherical) shape in 2-D (3-D) is most efficient due to the physics regarding sound field reproduction.
In a second aspect, the present invention broadly consists in an audio device for driving multiple steerable loudspeakers to reproduce a spatial sound field in a sound control region, each loudspeaker having a plurality of different individual directional response channels being controlled by respective speaker input signals to generate sound waves emanating from the loudspeaker with a desired overall directional response created by a combination of the individual directional responses, and where the loudspeakers are located about a sound control region in a room having at least one sound reflective surface, the device comprising: an input interface for receiving input spatial audio signals representing a spatial sound field for reproduction in the sound control region; a filter module comprising filters that are configurable based on acoustic transfer function data representing the acoustic transfer functions measured in the sound control region from the individual directional responses of each of the loudspeakers at their respective locations in the room, and which filter the input spatial audio signals to generate speaker input signals for all the loudspeakers to generate sound waves with co-ordinated overall directional responses that combine together at the sound control region in the form of either direct sound or reflected sound from the reflective surface(s) of the room to reproduce the spatial sound field; and an output interface for connecting to all the loudspeakers and for sending the speaker input signals to the loudspeakers.
In one form, the input interface may be configured to receive input spatial audio signals in an ambisonics-encoded surround format for direct filtering by the filters of the filter module to generate the speaker input signals for the loudspeakers.
In another form, the input interface may be configured to receive input spatial audio signals in a non-ambisonics surround format and which further comprises a converter that is configured to convert the non-ambisonics input signals into an ambisonics surround format for subsequent filtering by the filters of the filter module to generate the speaker input signals for the loudspeakers.
Preferably, the device may be switchable between a configuration mode in which the device configures the filters of the filter module for the room and a playback mode in which the device processes the input spatial audio signals for reproduction of the spatial sound field using the loudspeakers.
Preferably, the device may further comprise a configuration module that is arranged to automatically configure the filters of the filter module in the configuration mode based on input acoustic transfer function data for the room that is measured by a sound field recording system.
Preferably, the input acoustic transfer function data for the room may be measured by a sound field recording system comprising a microphone array located in the sound control region and the acoustic transfer function data represents the acoustic transfer functions measured by the microphone array in response to test signals generated by each of the loudspeakers for each of their directional responses.
Preferably, the configuration module may receive raw measured acoustic transfer function data from the sound field recording system and converts it into an ambisonics representation of the acoustic transfer function data which is used to configure the filters of the filter module.
Preferably, the filters of the filter module may be ambisonics loudspeaker filters.
The second aspect of the invention may have any one or more of the features mentioned in respect of the first aspect of the invention.
The phrase “direct sound” in this specification and claims is intended to mean sound waves propagating directly from the loudspeaker into the sound control region without reflection of any reflective surfaces.
The phrase “reflected sound” in this specification and claims is intended to mean sound waves propagating indirectly from the loudspeaker into the sound control region after being reflected off one or more reflective surfaces, whether 1st order reflections, 2nd order reflections, or higher order reflections, such that the sound waves appear to be arriving from virtual sound sources not corresponding to the loudspeakers.
The term “comprising” as used in this specification and claims means “consisting at least in part of”. When interpreting each statement in this specification and claims that includes the term “comprising”, features other than that or those prefaced by the term may also be present. Related terms such as “comprise” and “comprises” are to be interpreted in the same manner.
As used herein the term “and/or” means “and” or “or”, or both.
As used herein “(s)” following a noun means the plural and/or singular forms of the noun.
The invention consists in the foregoing and also envisages constructions of which the following gives examples only.
Preferred embodiments of the invention will be described by way of example only and with reference to the drawings, in which:
The present invention relates to a surround sound system for reproducing a spatial sound field in a room, typically for domestic home entertainment systems. The surround sound system is scalable to suit rooms of varying size and shape. Typically the room is substantially enclosed by a floor and ceiling, and comprises at least one but preferably multiple sound reflective or reverberant surfaces, typically provided by a wall(s) defining the room or other vertical surface adjoining the floor and ceiling. The levels of reverberation are measured by the critical reverberation distance which represents the distance from a source at which the reverberant and direct sound energies are equal. In an average living room or bedroom, this distance is typically 50 cm to 1 meter. Any further than the critical reverberation distance, sound energy is dominated by the reverberation.
In brief, the surround sound system is configured to generate spatial or surround sound by creating the impression that sound is coming from one or more intended directions. Referring to
The surround sound system could be implemented with a 2-D spatial sound field reproduction or a more complex 3-D sound field reproduction. The example embodiments of the surround sound system to be described focus on the 2-D implementation with the sound control region located in a substantially horizontal plane in space within the room environment and with the array of loudspeakers located in substantially the same plane in space, but the design modifications required for providing a 3-D implementation will also be discussed, which may involve a spherical sound control region and employing loudspeakers in locations on the ceiling and floors.
More specifically, in this specification unless the context suggests otherwise, 2-D spatial sound field reproduction is intended to relate to reproduction of the spatial sound in a 2-D sound control region, typically circular, which may have a desired predefined height or thickness vertically, and in which the surround sound system may typically comprises a circular array of loudspeakers surrounding the 2-D sound control region and which are arranged to propagate sound waves horizontally into the sound control region. The thickness of the 2-D sound control region may be determined by the loudspeaker vertical dimensions, or whether the loudspeakers are vertical line arrays or electrostatic loudspeakers that are capable of propogating sound waves horizontally toward the sound control region over a vertical range corresponding to the thickness of the 2-D sound control region. In this specification, unless the context suggests otherwise, 3-D spatial sound field reproduction is intended to relate to the spatial sound in a 3-D sound control region, typically a spherical region, and in which the surround sound system may comprise a spherical array of loudspeakers surrounding the 3-D sound control region and which are oriented or configured to propagate sound waves into the 3-D sound control region at any desired elevation angle, whether horizontal, vertical or any other angle.
In this embodiment, the control unit 14 has two modes of operation, a configuration mode and a playback mode. The configuration mode must be operated at least once before the playback mode can operate effectively. During set-up of the surround sound system, the configuration mode is initiated once all the loudspeakers are positioned about the sound control region in the room. The configuration mode customises the performance of the system to the loudspeaker layout and reverberance properties of the room so as to configures the responses of the loudspeakers to exploit the natural reverabaration in the room, and to use both the direct sound path and available reverberant reflections to reproduce the spatial sound field represented by an input spatial audio signal when in playback mode. Once configured, the system can be switched into playback mode for sound field reproduction. The system typically remains in playback mode until the loudspeaker positions are altered or the room reverberation properties changed in any way, in which case the configuration mode is typically re-initiated to re-calibrate the system for the new set-up or environment.
By way of example only, the system is capable of reproducing a full ambisonics sound field, but also emulating or reproducing other spatial sound signal formats, including Dolby surround and others. The surround sound system may be a stand-alone system that receives the input spatial audio signals 16 from another audio playback device, Personal Computer, or home theatre or entertainment system, or may be integrated as a component or functionality of such systems or devices.
The various components and mode operations of the surround sound system will now be individually described in more detail.
Referring to
In this embodiment, the input spatial audio signals 16 containing the spatial audio information is delivered to the control unit 14 as several input sound channels. By way of example, it may be composed of (i) ambisonically-encoded sound information, (ii) spatial information on the phantom source location(s) from which each sound channel will be played, or (iii) one of a variety of surround-formatted signals. By way of example only, the surround multi-format signals could include: stereo, Dolby Digital™, DTS Digital Surround™, THX Surround EX, DTS-ES and others.
In this embodiment, the control unit 14 is configured to receive either an ambisonically-encoded input signals 16a or one or more other formats of surround-encoded input signals 16b. The ambisoncially-encoded 16a input signals are filtered directly by the filters 17, while other format signals 16b are first processed by an ambisonics converter 18 and converted into an ambisonics format for subsequent processing by the filters 17. It will be appreciated that other embodiments of the control unit need not necessarily provide this multi-format input capability and may provide only one format of input signal if desired. In operation, the central control unit 14 processes and delivers each of the excitation input signals to the directional response components of each smart loudspeaker unit 12 for playback of and reproduction of the spatial sound field.
As previously discussed, the pre-configured filters 17 are configured or customised for the arrangement of loudspeakers 12 and room reverberation characteristics in the configuration mode. This is achieved by measuring acoustic transfer functions for each of the loudspeaker directional responses in the sound control region, which will be explained in further detail later. The signal processing performed by the central control unit 14 and the storage of acoustic transfer functions in the ambisonically-encoded spatial sound format will be described in further detail below.
As shown, the control unit 14 also comprises a configuration module in the form of a surround sound processor 19 that is configured to measure the acoustic transfer functions to the sound control region at a number of frequencies in the configuration mode of the system and then configure the filters 17 based on those measured acoustic transfer functions. As shown in
The ambisonics input signal 16, surround sound processor 19, ambisonics converter 18, and ambisoncics loudspeaker filters 17 will each be described in further detail below.
2.1 Ambisonics Input Signal
The central control unit 14 requires information regarding the spatial placement of the sound. Ambisonics pertains to the representation of a spatial sound field. Ambisonics has both 2-D and 3-D versions. The B-format recording is one of the earliest realizations of ambisonics, which records the sound pressure and 3 components of velocity at a point in space, then reproduces the sound field using an array of loudspeakers [9]. For 2-D reproduction, only two components of velocity are measured. The ambisonics B-format thus consists of 3 signals in 2-D (pressure plus two components of velocity) and 4 signals in 3-D (pressure plus three velocity components). This sound field is reproduced accurately over a large area only at low frequencies. Since the area of accurate reproduction reduces with frequency, this spatial sound reproduction is inadequate over much of the audible frequency range. For a disc-shaped (2-D) or spherical control region (3-D) the radius for accurate reproduction is only R=sv/2πf=55 mm at 1 kHz where sv is the speed of sound.
For sound field reconstruction over a larger area, one may use Higher Order Ambisonics (HOA), which is adopted in the surround sound system of the invention. In HOA, the sound field at each point (r,φ) over a circular region at frequency f can be written in terms of the ambisonics expansion about the origin:
where Jn(·) is the Bessel function of order n, βn(f) is the 2-D ambisonics coefficient at frequency f, k=2πf/sv is the wave number and N is the order of the ambisonics field related to the radius of the circular region by R=Nsv/2πf (For a B-format recording, N=1). We record the sound field by measuring the coefficients over a finite range n=−N, . . . , N producing the Nth order ambisonics signal set. One requires at least 2N+1 drivers to reproduce the Nth order HOA in 2-D.
The sound field at each point (r,θ,φ) over a 3D spherical region can be written in terms of the ambisonics expansion about the origin:
where jq(·) is the spherical Bessel function of order n, Yqp(·) is the spherical harmonic function and βqp(f) is the 3-D ambisonics coefficient. One requires at least (N+1)2 drivers to reproduce the Nth order HOA in 3-D.
There are equivalent ambisonic representations to the complex angular functions einφ (2-D) or Yqp(θ,φ) (3-D) which are real. Either the real or the complex functions could be used in the surround sound system of the invention. Real representations have implementation advantages but are easy to obtain from the complex functions [11].
Alternatively to ambisonics, the input audio signal spatial information delivered to the central control unit 14 could consist of a number of sound channels, each for several phantom source, each channel additionally having the following specified:
There are standard equations for converting such spatial sound information into an ambisonics format. Such equations shall be used to reconstruct the sound fields up to Nth order ambisonics for the loudspeaker location of a Dolby Surround, DTS or other commercial surround system.
2.2 Surround Sound Processor and Configuration Mode
As mentioned above, the surround sound processor 19 of the control unit 14 is operable to receive and process acoustic transfer function data 21 representing the acoustic transfer functions measured during the configuration mode by the microphone array 20. At a general level, to determine the acoustic transfer functions in the room, a number of test signals are played out of each smart loudspeaker, and the response recorded by the central control unit 14 using a microphone array.
For determining each of the acoustic transfer functions, a test signal 22 is generated and directed to each channel of each smart loudspeaker. Each channel of the loudspeaker generates a different directional response. The impulse response to each microphone in the microphone array is then measured. The test signal used may be a pulse signal, but more practically a wideband chirp or Maximum Length Sequence signal may be used. The filters 17 can then be configured in the frequency domain, using just the positive frequencies, so it is possible to measure the complex ambisonics coefficients of the acoustic transfer functions. Ambisonics is an efficient means of storing the acoustic transfer function for each channel of each smart loudspeaker at a number of frequencies. This control unit 14 stores the acoustic transfer function data in the form of the ambisonic loudspeaker filters 17 after signal processing to be detailed below. In brief, the surround sound processor 19 takes the measured acoustic transfer function data, applies FFT and mode weighting matrices, then does a matrix inversion before it stores the data into the ambisonic loudspeaker filters 17.
More particularly, the surround sound processor 19 is configured to receive and convert the raw microphone array acoustic transfer function data at each frequency into the (ambisonic) modal decomposition of the acoustic transfer functions in equations (3) and (5) below, in 2-D by using a FFT matrix 23 followed by a phase mode weighting matrix 24 dependent on the array radius and type of housing [4] or in 3-D by using a spherical harmonic transform matrix followed by a 3-D mode weighting matrix [14]. The surround sound processor is then arranged to configure the ambisonic loudspeaker filters 17 based on the measured and processed acoustic transfer function coefficients, and which is explained in further detail below.
The use of a microphone array for sound field recording is known by those skilled in the art. Any suitable microphone array design may be used that is capable of measuring the acoustic transfer functions from each loudspeaker to any point in the sound control region [1-4]. A 2-D implementation may use a uniform circular array geometry 20 as shown in
As mentioned, the computation and configuration of the ambisonic loudspeaker filters 17 for sound reproduction is implemented within the Surround Sound Processor 19. This process for the 2-D implementation is first explained, followed by the 3-D implementation. It is desired to reproduce a number of ambisonic sound fields using a set of L smart loudspeakers.
For the 2-D implementation, consider a sound field with expansion about an origin given by ambisonics expansion in equation (1). The ambisonics coefficients of the desired sound field are βn(f) expressed in the frequency domain. The control unit 14 requires a set of acoustic transfer functions for each loudspeaker. The acoustic transfer functions are efficiently stored as a set of ambisonically-encoded modal coefficients αn(l,m|f) defined in terms of the sound field created by the mth directional response of each loudspeaker l:
The coefficients αn(l,m|f) are measured in the configuration mode of operation at the intended listening position with aid of the microphone array 20. A total of (2M+1)L sets of 2N+1 coefficients are produced.
As mentioned, the surround sound processor 19 of the central control unit 14 determines the loudspeaker filters to be applied to the spatial audio signals based on the measured acoustic transfer functions. In a preferred embodiment, the loudspeaker filters are designed to reconstruct the nth spatial sound mode Jn(kr)einφ. We determine the loudspeaker filters Gn(l,m|f) to recreate each nth spatial mode as follows: The sound pressure resulting in the room from the loudspeaker weights for creating the nth mode {Gn(l,m|f): m=1, . . . , 2M+1, l=1 . . . L} is:
Substituting in equation (3), we determine an equation for determining each loudspeaker filter:
which by orthogonality of complex exponentials is satisfied if the following set of equations are satisfied:
for n′=−N, . . . , N. This set of 2N+1 equations can be written in matrix-vector form:
A(f)gn(f)=en,
where [A(f)]n+N+1,(l−1)(2M+1)+m=αn(l,m|f), [gn(f)](l−1)(2M+1)=Gn(l,m|f) and en is an 2N+1-long vector where element n+N+1 is one and all other elements are zero. Here [M]ij denotes the element in the ith row an jth column in matrix M whilst [v]i denotes the ith element of vector v. Vector gn(f) contains the L(2M+1) loudspeaker filter weights at frequency f to apply to the configurable loudspeaker channels to create the spatial mode corresponding to the nth ambisonic coefficient. As a result, a matrix G(f) [g−N(f), g−N+1(f), . . . , gN(f)], whose 2N+1 columns are the loudspeaker weight vectors for creating the ambisonic spatial sounds at frequency f up to order N, can be determined by taking the regularized pseudo-inverse of A(f) through the Tikhonov-regularized least squares. The matrix A(f) is long, since a robust solution would entail using more drivers, L(2M+1), than the 2N+1 reproducible ambisonic channels. As a result the solution is:
G(f)=A(f)H[A(f)A(f)H+λI]−1 (4)
where λ is a single regularization parameter. The parameter λ may either be tuneable or have a fixed value selected in the device.
The required filters to create the 2-D ambisonics spatial sound field are shown to be related to the 2M+1 acoustic transfer function coefficients for each of the L configurable loudspeakers. There are L(2M+1) acoustic transfer functions for each mode. The Surround Sound Processor 19 hence determines the ambisonics loudspeaker filters directly from the measured acoustic transfer function coefficients.
The approach presented here represents a frequency-domain approach, where the output is a collection of loudspeaker weights at a number of frequencies. This approach culminates in a time-domain approach, where the output is a collection of time-domain filters. The solutions may be calculated at each frequency, and the inverse FFT used to produce the required digital filter for filtering the nth ambisonics signal for the mth mode of the lth loudspeaker.
In a 3-D implementation, the desired spatial sound field can be written as equation (2) where βqp(f) is now an ambisonics coefficient of the desired sound field. The acoustic transfer functions are efficiently stored as a set of ambisonically-encoded modal coefficients αqp(l,m|f) defined in terms of the sound field created by the mth directional response of each loudspeaker l:
In a preferred embodiment, the loudspeaker filters are designed to reconstruct the (p,q)th ambisonic spatial sound mode jq(kr)Yqp(φ,θ). We determine the loudspeaker weights Gqp(l,m|f) to recreate each spatial mode (p,q) at frequency f as follows. The sound pressure resulting in the room from loudspeaker weights is:
Substituting in equation (5), we obtain an equations for determining the (p,q)th loudspeaker filter
which by orthogonality of spherical harmonics is satisfied if the following set of equations are true:
for {(p′,q′): q′=0, 1, . . . , N, p′=−q′, . . . , q′}. The set of (N+1)2 equations for each (p,q) can be written in matrix-vector form as:
A(f)gqp(f)=eqp,
where [A(f)]p
The required filters to create the (p,q)th 3-D ambisonics spatial sound field are again related to the (M+1)2 acoustic transfer function coefficients for each of the L smart loudspeakers corresponding to the same mode (p,q). There are L(M+1)2 acoustic transfer functions for each mode.
2.3 Ambisonics Loudspeaker Filters
As mentioned above, the ambisonics loudspeaker filters 17 of the control unit 14 are configured for the room during the configuration mode prior to switching to the playback mode of the surround sound system. The filters may be digital filters, such as Finite Impulse Response (FIR) filters for example. The ambisonics loudspeaker filters 17 apply the appropriate filtering to construct the appropriate spatial sound field from each ambisonics input signal channel in playback mode shown in
In the 2-D embodiment of the system, the sound field represented by coefficients {βn(f): n=−N . . . N} is reproduced using several smart loudspeakers 12, each of which is capable of generating 2M+1 polar responses, M being the order of the directional response. In this embodiment, each configurable loudspeaker may contain from M=1 to 4, although higher order directional responses, e.g. up to 20th order or higher still may be required for higher operating frequencies. As shown in
By way of example only, to reconstruct sounds at 1 kHz (2 kHz) in a disc of diameter 60 cm (30 cm) sound control region, at least an ambisonics order of N=6 is required. The numbers of temporal loudspeaker filters for any conceivable 6th order 2-D ambisonics reproduction system are: 156≦L (2N+1) (2M+1)≦936 for L=4 to 8 configurable loudspeakers, and where M=1 to 4 in this embodiment, although it will be appreciated that the limits will alter if higher order loudspeakers are employed. More loudspeaker filters are required if the desire is to increase the size of the reproduction region beyond what is mentioned here.
2.4 Ambisonics Converter
In the embodiment shown in
In one embodiment, the Ambisonics Converter 18 is used for converting Dolby 5.1 surround signals 16b into ambisonics coefficients 18a to generate phantom sources positioned in the standard five loudspeaker ITU geometry used in Dolby Digital and DTS Digital Surround. In an alternative embodiment, the Ambisonics Converter 18 could also support stereo sound or the seven loudspeaker layouts of THX Surround EX and DTS-ES where the loudspeaker locations are different. The converter 18 makes the surround sound system downward compatibility with currently-available technologies.
By way of example, we show one possible method of converting these surround sound formats into an ambisonic format given the desired loudspeaker locations. For an acoustic monopole in 3-D, the sound pressure at point x=(r,θ,φ) truncated to Nth order ambisonics is:
where y=(rs,θs,φs) is the position of the monopole source and S(f) is the transmitted sound signal. For an acoustic monopole in 2-D, the sound pressure at point x=(r,φ) for a monopole source located at y=(rs,φs) the Nth order ambisonic reconstruction of the sound pressure is:
where Hn(2)(·) is the Hankel function of the second kind of order n. The ambisonics coefficients of an acoustic monopole are hence βqp(f)=ikhq(2)(krs)[Yqp(θs,φs)]*(3-D embodiment) and βn(f)=Hn(2)(krs)e−inφ
3.1 Design of Loudspeaker
Each loudspeaker 12 is capable of creating a number of configurable directional responses over a number of frequencies, and may preferably have the capability of steerability of the beam pattern in 360° in the 2-D implementation. Each smart loudspeaker 12 is driven by several speaker input signals 13, each signal line drives a separate loudspeaker directional response. The loudspeakers 12 may provide onboard amplification to each driving signal, or alternatively the amplification may be provided in the central control unit or other amplifier module(s), whether integrated with the central control unit or each loudspeaker or provided as a separate component.
As shown in
As mentioned, a series of D amplifiers 26 may be provided for magnifying the signals to volume levels appropriate for playback. The amplified signals are each delivered to a loudspeaker (driver) co-located in common housing. In this embodiment, the housing is compact and the driver 25 geometry in each loudspeaker 12 is chosen to generate directional patterns over a range of directions. A circular driver geometry is shown in
The number of drivers and input channels 13 for the loudspeakers 12 may vary depending on the surround sound system playback requirements. For the surround sound system to exploit room reflections, it is generally required for each configurable loudspeaker to be able to create at least a M=1st order directivity pattern, and preferably up to 4th order.
The loudspeakers 12 create directional responses up to Mth order using a small number D of drivers (D≧2M+1 in 2-D and D≧(M+1)2 in 3-D). The 2-D implementation of the smart loudspeaker might include (i) constructing the 2M+1 phase mode directional responses {eimφ: m=−M, . . . , M}, (ii) constructing an omni-directional response, as well as each of the directional responses cos(mφ) and sin(mφ) for m=1, 2, . . . , M. For a 3-D implementation, the smart loudspeaker could construct an omni-directional response, as well as the real parts {Re[Ynm(θ,φ)]: m=0 . . . n, n=1 . . . M} and imaginary parts {Im[Ynm(θ,φ)]: m=1 . . . n, n=1 . . . M} of the spherical harmonic functions. The Loudspeaker Beamspace Matrix 27 and the geometric arrangement of the drivers within the housing of the configurable loudspeaker unit 12 are selected to create such directional responses over a wide range of frequencies. These design aspects are further described below.
The physical layout of the drivers within the loudspeaker 12 will now be described. The far-field directivity pattern Dl(φ|f) of loudspeaker l at frequency f can be written as the phase mode expansion:
where αn(l|f) are the weighting coefficients for the nth order phase mode. Each directional loudspeaker is realized by arranging a number D of monopoles drivers into a uniform circular array of radius r. To ensure loudspeaker responses up to Nth order are obtainable, one designs each monopole array choosing r and D as follows:
This scheme ensures monopoles are spaced λ/2 or less apart to avoid spatial aliasing at frequency f, corresponding to the lowest frequency in the operating frequency of the surround sound system. The array design may be constructed by housing the D drivers inside a cylindrical loudspeaker box. The driver weights are then chosen according to regularized least squares to suit the sound field reproduction problem. Typically, the audio operating frequency range of the surround sound system is preferably in the range of 60 Hz-12 kHz, more preferably 30 Hz-20 kHz.
As discussed, the beamformer module of each loudspeaker 12 may be in the form of a beamspace matrix. Each loudspeaker is designed to generate the 2M+1 directional responses (2-D implementation) or (M+1)2 responses (3-D implementation) up to order M, using D drivers. By way of example, the following illustrates the design for acoustic monopole drivers in free-space in one embodiment of the loudspeaker design. In alternative 2-D embodiments, the drivers are mounted onto the equator of a hard cylinder or sphere. Suppose each monopole d of a directional loudspeaker at frequency f is excited by loudspeaker weight bmd(f) where m=−, . . . , M and d=1, 2, . . . , D. To choose the loudspeaker weights to construct the nth phase mode in the far-field, it is necessary to match the directivity pattern eimφ across the continuous angular range φε[0,2π]:
where θd=[cos θd, sin θd]T, θm is the orientation angle of monopole m and φ=[cos φ, sin φ]T. If the loudspeaker vector for the D element array to construct the mth order phase mode is bm=[bm1, bm2, . . . , bmD]T then bm can be designed by matching the directivity pattern at Q angles {φ1, φ2, . . . , φQ}:
Ebm=pm
where [pn]q=eipφ
B=E+P
where P=[p−M, . . . , pm] and E+=(EHE)−1EH is the pseudo-inverse of E. The matrix B for each loudspeaker transforms the 2M+1 phase mode weights into D driver weights.
The preferred directional responses for the channels of the loudspeakers are an omnidirectional pattern, cos mθ patterns and sin mθ patterns, (for m up to order M) are preferred. However, also acceptable are the phase mode responses eimθ (for m equalling −M up to M).
3.2 Physical Arrangement of Loudspeakers in Room
Typically, the surround sound system will function with L=4 to 8 configurable loudspeakers 12, although additional loudspeakers may increase performance of the system in certain environments.
In preferred embodiments, the room 5 is equally divided or segmented radially about the origin 6 at the center of the sound control region into loudspeaker location regions L1, L2, . . . LL, where L=the number of loudspeakers in the surround sound system. A loudspeaker is located at any location within its respective loudspeaker location region, such that there is one loudspeaker per loudspeaker location region. Each loudspeaker location region is defined to extend between a pair of dotted radii boundary lines B1, B2, . . . BL that extend outwardly from the origin of the sound control region. The angular distance θB between each pair of radii boundary lines is equal and corresponds to 360°/L, where L is the number of loudspeakers. In these preferred embodiments, additionally the loudspeakers are located at spaced-apart minimum distances from each other, adjacent walls, and the perimeter of the sound control region by the conditions lspkr, lwall, and lcontrol, which are further discussed below.
In
In
In
As shown in
One metric for suitability of a particular loudspeaker array configuration is the range of directions in which the image-sources are positioned. By way of example,
Statistical room acoustics, where the reverberant sound field is modelled as diffuse, would dictate that for the acoustic transfer functions at different loudspeaker locations to be uncorrelated and hence sufficiently different from each other, the loudspeakers must be located at least half a wavelength λ/2 apart. However at low frequencies, the surround system will tend to control individual room modes. The boundary between the statistical and modal descriptions of room acoustics is given by the Schroeder frequency, which is given by fS=2000√{square root over (T60/V)} where T60 is the standard room reverberation time and V is the room volume. Below the Schroeder frequency, the acoustic transfer functions become completely correlated. Hence lspkr=λS/2 and lwall=λS/4 are chosen using λS=sv/fS to ensure the loudspeaker acoustic transfer functions are uncorrelated and hence sufficiently different down to as low a frequency as possible. By way of example, in a living room of dimensions 5 m×4 m×2.5 m with a typical room reverberation time of 500 msec, the Schroeder frequency is 200 Hz. Using the above criteria, the loudspeakers should be spaced at least lspkr=86 cm apart and lwall=43 cm away from walls.
A reasonable distance of loudspeakers from the centre of the sound control region lcontrol is required to help ensure that the direct sound is not large in comparison to the sound of a reverberant reflection. This condition helps ensure exploiting a reflection for surround sound is robust. The actual distance will depend on both the directivity of the array which is related to loudspeaker order M, and to a lesser extent the strength of wall reflections. Considerations for choosing lcontrol are elaborated on below.
In other embodiments, the geometrical arrangement of the loudspeakers may correspond to the ITU-R BS 775 5.1 Dolby Surround geometry if there are five loudspeakers employed, with a center speaker at 0° in front of the listener in the sound control region, left and right front surround speakers located at +/−22.5-30° and left and right rear surround speakers located at +/−90-110°. Additionally, if seven loudspeakers are employed, the Dolby Surround 7.1 geometry may be employed.
3.3 Number of Loudspeakers and Loudspeaker Order
The requirements on the number loudspeakers L and the directional loudspeaker order M are a function of the radius of the sound control region R and the acoustic frequency f and can be approximately determined from the rule of thumb:
To determine the directional loudspeaker order M as a function of R, f and L, this equation can be rearranged to obtain:
where ┌x┐ is the integer ceiling function of x.
To create a control region of a constant size with frequency, the directional loudspeaker order must be stepped up progressively at pre-determined frequency thresholds. By way of example, for a sound control region of radius R=0.2 m, the frequency thresholds for typical choices of the numbers of loudspeakers 12 are shown in Table 1. This table shows that the requirements on loudspeaker order can be reduced by increasing the numbers of loudspeakers 12.
In preferred embodiments, the control unit of the surround sound system is configured to automatically step-up the order of the directivity patterns of the overall directional responses of the loudspeakers as the frequency of the spatial sound field represented by the input spatial audio signals increases to thereby maintain a substantially constant size of sound control region. As shown by the above example, the control unit is preferably configured to step-up the order of the directivity pattern at predetermined frequency thresholds that are predetermined and calculated based on the number of loudspeakers and the desired size of the sound control region.
3.4 Preferable Sound Control Region Size
The diameter 2R of the sound control region cannot be any smaller than the size of the listener's head, and would preferably include both the head and shoulders. On average, the diameter of a human head is accepted to be 0.175 m. Due to the heavy requirements on number of drivers required to perform sound reproduction at high frequencies, the sound control region diameter would typically be no larger than 1 m in most commercial applications, although larger control regions could be provided for as will be appreciated.
3.5 Preferable Room Conditions
The preferable room conditions of the surround sound system are a function of the strength of wall reflections, and the relative lengths of the paths of direct propagation and the reflected propagation path, from loudspeakers 12 to the sound control region. To exploit a reflection, due to the longer propagation distances and the energy absorbed by each wall reflection, the sound directed toward the wall will have to be boosted by the loudspeaker 12 over the levels required for direct sound propagation.
Strong boosting of the sound directed toward the wall reflection however is ill-advised, as such boosting increases the average sound energy levels outside the sound control region [5]. These sound levels may be perceived as unpleasant to a listener standing outside. The external sound levels can be reduced to acceptable levels by appropriate choice of Tikhonov regularization parameter. For good system performance, room conditions must hence be able to ensure the sound energy levels outside are not required to be made significantly larger than those inside the sound control region.
By way of example consider an room with identical reflecting walls of sound energy absorption coefficient α. Define lcontrol as the distance of loudspeakers from the sound control region and lmfp=4V/S as the mean free path where V is room volume and S is total room surface area. For an nth order reflection, the propagation distance to the control region is approximately n lmfp. For 2-D line sources, the loudspeakers energy will have attenuated down to 10 log10(lcontrol/n lmfp) of the direct sound field energy due to the propagation distance losses, and 10n log10(1−α) due to wall energy absorption. Reflections must hence be boosted by the loudspeaker to counteract this level of attenuation:
This equation assumes specular reflection only and does not include air absorption losses which are assumed small. For loudspeakers lcontrol=1 m away from the sound control region in the 5 m×4 m×2.5 m room (so that lmfp=2.4 m) with walls having 50% sound absorption, to exploit 1st, 2nd and 3rd order reflections, these reflections must be boosted by 6.7 dB, 13 dB and 18 dB respectively, with the more significant contributor of the attenuation being the greater distance of the higher order reflections from the sound control region. The control unit is configured to boost or amplify the signals relating to the reflected sound to account for wall attenuation. We note that approximate line sources can be built using vertical line arrays or electrostatic loudspeakers. Similar analyses can be applied for 3-D sources, where the dependence of propagation loss on distance l is proportional to 20 log10 l instead.
Typically, the system preferably exploits 1st, 2nd and 3rd order reflections in rooms with a wall energy absorption coefficient no greater than 75%, and preferably less than 50% to ensure higher order reflections do not require excessive boosting. Due to the distance and wall reflection attenuation aspects, the surround sound system would typically not be configured to exploit reflections beyond 3rd order.
Due to the difference in lengths of the propagation paths between the direct sound and higher order reflections, loudspeakers should typically be spaced at least lcontrol=1 m away from the center of the sound control region, and preferably more than 1.5 m.
Embodiments of the surround sound system may have the following applications:
The system provides these benefits through a surround sound system that employs the use of multiple configurable directional loudspeakers to exploit reverberant reflection in the performing of surround sound. The system employs a sparse array geometry of loudspeakers, with loudspeakers located near the edges or corners of the room, for exploiting the reverberant reflection. The system employs a smaller number of loudspeakers than would be required by a traditional higher order ambisonics system. Further, the surround sound system creates the impression of sound originating from a wall reflection utilising to some extent all loudspeakers, and to not only create the spatial sound impression but also utilise the loudspeakers to cancel at least some of the unwanted reverberation caused by other sound reflections, as the system performs sound field reproduction by means of reverberant compensation.
A first experimental example of the surround sound system will be described by way of example and is not intended to be limiting. Like reference numbers in the drawing refer to the same or similar components. In this experimental example of the surround sound system it is shown that using a small number of directionally-controlled loudspeakers, a sound field may be accurately reproduced in a reverberant room. The goal of surround sound is to reproduce a sound field within a control region. Using constructive and destructive interference from the waves emitted from a set of directional loudspeakers, sound field reproduction can be used to create an arbitrary sound field in the control region.
A common objective in surround sound is to place one or more phantom sources around the listener. To place a phantom source at any intended orientation, one would ideally distribute adequate loudspeakers evenly around the listener, with sufficient numbers to avoid spatial aliasing. One such geometry is the uniform circular array (UCA). To meet aliasing requirements in 2-D, at least 2kR+1 loudspeakers are required [19]. However, neither this loudspeaker geometry nor the large numbers of loudspeaker are practical, as both aspects demand a large amount of physical space in the room which carries a low spouse-acceptance-factor.
The surround sound system of the invention reduces the heavy requirements on numbers and arrangement of loudspeakers by using a loudspeaker configuration which exploits room reverberation.
Referring to
Through exploring the performance of the corner array shown in
In this experimental example, we consider reproducing the sound field over a volume of space with a small number L of steerable directional loudspeakers 12. Each configurable directional loudspeaker is realized using an identical array of 2-D monopole elements, so that reverberation can be easily simulated using the image-source method [13]. Here the loudspeakers synthesise directional responses up to approximately M=3rd order. In this experiment, we restrict attention to 2-D reproduction in a room using vertical line sources. The purpose of the steerable loudspeaker approach is to generate additional phantom image directions by creating beams which bounce off reflective walls. Quantitative features of the reverberant sound field are accurately modelled by the image-source method for the case of specular reflection. By exploiting specular reflections, we can improve performance in reverberant environments.
We first overview the pressure matching approach to sound field reproduction. We then describe the approach to modelling the directional loudspeaker.
5.1 Pressure Matching
In the pressure matching approach, one reproduces a desired sound field by matching the pressure at a finite number of points within the sound control region. We shall refer to these points as the matching points. The control region is a circular 2-D region of radius R. To reproduce the desired pressure field Ps(x;f) over the control region using the L directional loudspeakers of D 2-D monopole elements, one needs to satisfy the equation at every point x in the sound control region:
where H(x|vld,f) is the acoustic transfer function between a monopole driver at yld and a point x. Pressure matching is performed over a dense grid of Q′ matching points {x1, . . . , xQ′} located within the control region. The set of equations required to be satisfied can be manipulated into the matrix-vector form
Hg=pd
where [H]q(Dl+d)=H(xq|yld,f) is a matrix of acoustic transfer functions, [g]Dl+d=Gld(f) is a vector of loudspeaker weights and [pd]q=Pd(xq|f) is a vector of desired pressures at the matching points. The loudspeaker weights g required to achieve a small mean square error robustly can be calculated through the regularized least squares solution:
g=[HHH+λI]−1HHpd (6)
where λ is the Tikhonov regularization parameter. A class of desired pressure fields that shall be reproduced here is the 2-D phantom monopole source:
Pd(x|f)=P0H0(2)(k∥x−Rsφs∥),
where Rs is phantom source radius, φs=[cos φs,sin φ2]T, φs is the orientation angle of the phantom source and P0 is a pressure amplitude constant.
For accurate sound field reproduction over a circular 2-D region of radius R, the number of monopoles required at wavenumber k [15] is:
L′=2kR+1 (7)
This number corresponds to the number of spatial modes active within the control region.
5.2 Directional Loudspeaker Design
A directional loudspeaker can be modelled with an Mth order directivity pattern. The far-field directivity pattern Dl(φ|f) at frequency f can be written as the phase mode expansion:
where αml(f) are the weighting coefficients for the mth order phase mode. Each directional loudspeaker is realized by arranging a number D of monopoles drivers into a uniform circular array of radius r. To ensure loudspeaker responses up to Mth order are obtainable, one designs each monopole array choosing r=M/k and D≧2M+1 as described above. Here we ensure the directional loudspeakers are designed to achieve second order directivity responses. The monopole weights are then chosen according to regularized least squares to suit the sound field reproduction problem.
The near-field directivity pattern Dl(φ|f) of each configurable directional loudspeaker l that results from the above pressure matching design is:
where ρ is the distance from the centre of the uniform circular array of the loudspeakers, φ the angle made with the x-axis, φ=[cos φ, sin φ]T, φd=[cos φd, sin φd]T and φm is the orientation angle of each loudspeaker m.
5.3 Pressure Matching with a Uniform Circular Array
For comparison in this experiment, we shall also reproduce the sound field with L′=LD acoustic monopoles arranged into a uniform circular array. Matching the pressure over Q′ points inside the sound control region, the loudspeaker weights are again obtained through the regularized least squares solution in equation (6) where instead [H]ml=H(x|yl,f) is now the acoustic transfer function between a monopole at located at yl in the UCA and a point sensor at x.
5.4 On Robust Design
We briefly discuss aspects which contribute to the robustness of a surround sound system. The way the robustness is quantified is through the loudspeaker weight energy ∥g∥2. The white noise gain [17, p. 69], quantifies the ability of a loudspeaker array to suppress spatially uncorrelated noise in the source signal. The major errors such as those in the amplitude and phase of the acoustic transfer functions and loudspeaker position errors are nearly uncorrelated and affect the signal processing in a manner similar to spatially white noise [18]. As the loudspeaker weight energy is inversely proportional to the white noise gain, it provides a relative measure of the reaction to such errors.
We examine the factors affecting robustness with aid of the singular value decomposition (SVD). In the case L′≦M, the SVD of the acoustic transfer function matrix H can be written:
where un are the orthonormal output vectors of the sound fields reconstructible by H, vn are the orthonormal input vectors of loudspeaker weights and σn are the singular values of matrix H describing the strength of the sound field created by each loudspeaker weight vn. We shall assume singular values are ordered σ1>σ2> . . . >σL′. After substituting the SVD of H into equation (6), the loudspeaker weights can be shown to be:
where cn=unHpd is the projection of pd on the subspace of sound fields reconstructable by H.
A straight-forward way of improving robustness is to increase the Tikhonov regularization parameter λ. The loudspeaker weight energy can be shown to be:
which is inversely related to λ. It is largest if we choose a vector as the sound field g=uL′ with the smallest singular value, where loudspeaker weight energy is equal to σL′2(σL′2+λ)2. Increasing λ however reduces the size of the loudspeaker weight energy at the expense of performance.
In contrast, manipulating the acoustic environment's geometry so that the desired sound field pd projects onto only the reconstructable sound fields un having large singular values σn would also improve robustness. Robustness can be improved by:
As illustrated by the arrows 32 and 34 in
In the case of the array of directional loudspeakers, the loudspeaker weight energy includes a component attributable to the ease of realizing the directional patterns with the D monopole drivers. The measure hence relies on the directional loudspeaker being properly designed, which will be the case if the number and geometry of the monopoles are chosen correctly for the design frequencies.
5.5 Results and Discussion
In this experiment, we demonstrate typical performance of a surround sound system with L=4 smart loudspeakers and 8 drivers in each configurable loudspeaker simulating performance at 500 Hz. The loudspeakers 12 were arranged in a corner array in a room 5 as shown in
We compared performance of the corner array with a uniform circular array (UCA) in a 6.4×5 m room under different reverberant conditions (cases):
The array geometries being compared are summarized as:
The sound control region 11 was located at the centre of the room 5 with a radius of R=0.5 m. We positioned the loudspeakers of the corner array away from the walls to increase the range of directions that can be attained from low order reverberant reflections.
Room reverberation was simulated using a 2-D implementation of the image-source method [13], with acoustic transfer functions computed using:
where ξi denote the accumulated reflection coefficient for the ith image-source and yl(i) the position of the ith image-source of monopole l, truncating the impulse responses to the T30 reverberation time. The T30 reverberation times are 530 msec and 100 msec for reverberant rooms 3 and 4 respectively. Sound field reproduction was carried out using the regularized pressure matching in with Tikhonov regularisation parameter λ=0.1 to create a 2-D monopole phantom source at 2 m from the centre of the control region. Due to the symmetry in the room geometry, it was sufficient to pan the phantom source angle over a 90° angular range.
We compare the performance of the corner array with that of an UCA of 32 loudspeakers in reverberant room case 3. For a 0.5 m control region radius, only 11 monopoles are required by (7) at 500 Hz, so there are a number of additional degrees of freedom with which to perform the reproduction. These degrees of freedom are not wasted, as adding loudspeakers above the Nyquist sampling requirements improves the robustness.
The MSE reproduction performance of the corner array in several acoustic environments is shown in
Marked on
Higher order images also contribute to improving MSE performance. In
The level of performance is dependent upon the strength of reverberant reflections. Reducing the strength of reverberant reflections decreases performance. The dotted curve 46 in
In regard to beampatterns, the directional loudspeaker corner array performance is best when the phantom source is in-line with either a loudspeaker or a low order reflection. By way of example, phantom sources are placed in directions of D and R illustrated in
5.6 Summary
This experiment tested an approach to surround sound for exact sound field reproduction in a reverberant room by utilizing steerable loudspeakers with configurable directional responses. An array of four configurable steerable loudspeakers with roughly second order directivity was shown to possess a reproduction performance comparable with a much larger circular array of loudspeakers, by exploiting the wall reflections in a reverberant room. The level of performance was seen to be dependent on the strength of specular reflections. For optimal performance the room was seen to require strong wall reflections.
The pressure matching method in practise relies upon measurement of the acoustic transfer functions from each loudspeaker to a number of points in the sound control region. The approach must be made robust to error in these measurements and can be made robust through regularization.
A preliminary study of performance was presented using a corner array geometry for the smart loudspeakers. Other geometries also show potential, including a diamond and pentagon, and others. Although some geometries perform better than others for generating certain sound fields, the geometry studied here demonstrates the key features of using multiple steerable directional loudspeakers to exploit reverberation.
In this experimental example, a simulation of the surround sound system employing a 4 smart loudspeaker 12 corner array can generate a 1 kHz acoustic pulse propagating into the sound control region from an angle of 45 degrees.
To create the spatial sound pulse, the array first excites the bottom-left “smart” loudspeaker 12a at 0 msec which then bounces off the bottom wall at 4-8 msec. The bottom-right loudspeaker 12d adds some to the initial sound energy as it propagates past at 12 msec, before switching to the top-right loudspeaker 12c to contribute more energy to the wavefront at 16 msec. The wavefront then bounces off the right and top walls at 26 msec to again propagate past the top-right loudspeaker 12c which contributes more sound energy at 26-30 msec. After constructing the 45 degree wavefront in the sound control region at 34 msec, the four smart loudspeakers then antiphase the propagating sound to reduce its intensity and so ensure that no further reverberation reaches the control region.
The foregoing description of the invention includes preferred forms thereof. Modifications may be made thereto without departing from the scope of the invention as defined in the accompanying claims.
The following disclosure in the following documents is herein incorporated by reference.
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