EP1502475A1 - Verfahren und system zum repräsentieren eines schallfeldes - Google Patents

Verfahren und system zum repräsentieren eines schallfeldes

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Publication number
EP1502475A1
EP1502475A1 EP03749929A EP03749929A EP1502475A1 EP 1502475 A1 EP1502475 A1 EP 1502475A1 EP 03749929 A EP03749929 A EP 03749929A EP 03749929 A EP03749929 A EP 03749929A EP 1502475 A1 EP1502475 A1 EP 1502475A1
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EP
European Patent Office
Prior art keywords
representative
parameters
acquisition means
sensors
determining
Prior art date
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Granted
Application number
EP03749929A
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English (en)
French (fr)
Other versions
EP1502475B1 (de
EP1502475B8 (de
Inventor
Rémy BRUNO
Arnaud Laborie
Sébastian Montoya
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Trinnov Audio
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Individual
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Publication of EP1502475A1 publication Critical patent/EP1502475A1/de
Publication of EP1502475B1 publication Critical patent/EP1502475B1/de
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Publication of EP1502475B8 publication Critical patent/EP1502475B8/de
Anticipated expiration legal-status Critical
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Classifications

    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04RLOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R3/00Circuits for transducers, loudspeakers or microphones
    • H04R3/005Circuits for transducers, loudspeakers or microphones for combining the signals of two or more microphones
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04RLOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R3/00Circuits for transducers, loudspeakers or microphones
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04RLOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R3/00Circuits for transducers, loudspeakers or microphones
    • H04R3/04Circuits for transducers, loudspeakers or microphones for correcting frequency response

Definitions

  • the present invention relates to a method and a device for representing an acoustic field from signals delivered by acquisition means.
  • the methods and systems for acquiring and representing existing sound environments use models based on physically impracticable acquisition means, in particular as regards the electro-acoustic and / or structural characteristics of these acquisition means.
  • the acquisition means are, for example, made up of a set of elementary measurement elements or sensors arranged in specific places in space and having intrinsic electro-acoustic acquisition characteristics.
  • the systems grouped under the term "Ambisonic" only consider the directions of origin of the sounds relative to the center of the acquisition means formed by a plurality of elementary sensors, which leads to assimilate the acquisition means to a point microphone.
  • these systems represent the sound environment by modeling virtual sources whose angular distribution around the center theoretically makes it possible to obtain such a sound environment.
  • the object of the invention is to solve this problem by providing a method and a device delivering a representation of the acoustic field substantially independent of the characteristics of the acquisition means.
  • the subject of the present invention is a method of representing an acoustic field comprising a step of acquiring measurement signals delivered by acquisition means formed by one or more elementary sensors exposed to said acoustic field, characterized in that 'it comprises :
  • Said structural characteristics include at least position characteristics of said elementary sensors with respect to a predetermined reference point of said acquisition means;
  • - Said encoding filters are also representative of electro-acoustic characteristics of said acquisition means; - Said electro-acoustic characteristics include at least characteristics linked to the electro-acoustic capacities for intrinsic acquisition of said elementary sensors;
  • said step of determining the encoding filters comprises: a sub-step of determining a sampling matrix representative of the acquisition capacities of said acquisition means; a substep for determining an intercorrelation matrix representative of the resemblance between said measurement signals delivered by the elementary sensors forming said acquisition means; and a sub-step of determining an encoding matrix from said sampling matrix, from said intercorrelation matrix, and from a parameter representative of a desired compromise between the fidelity of representation of the acoustic field. and minimizing the background noise induced by the acquisition means, which matrix is representative of said encoding filters;
  • said step of determining the sampling matrix is carried out, for each of said elementary sensors forming said acquisition means, from:
  • step of determining the sampling matrix is further carried out on the basis of at least one of the parameters from:
  • parameters representative of the orientations of all or part of the sensors namely of their direction of maximum sensitivity;
  • said calibration step comprises, for at least one of said elementary sensors forming said acquisition means:
  • said calibration step further comprises:
  • a substep for transmitting a specific acoustic field to said at least one sensor said acquisition substep corresponding to the acquisition of the signals delivered by this sensor when exposed to said specific acoustic field;
  • said calibration step comprises a substep for receiving a finite number of signals representative of the electro-acoustic and structural characteristics of said sensors forming said acquisition means, which signals are directly used during said substep for determining the electro-acoustic and / or structural characteristics of said acquisition means;
  • the invention also relates to a computer program comprising program code instructions for executing the steps of the method, as described above when said program is executed on a computer.
  • the invention also relates to a mobile support of the type comprising at least one processing processor and a non-volatile memory element, characterized in that said memory comprises a program comprising code instructions for the execution of the steps of the method as described previously when said processor executes said program.
  • the invention also relates to a device for representing an acoustic field connectable to acquisition means formed by one or more elementary sensors delivering measurement signals when they are exposed to said acoustic field, characterized in that it comprises a module for processing the measurement signals by applying encoding filters representative of at least structural characteristics of said acquisition means to these measurement signals to deliver a signal which comprises a finite number of coefficients representative over time and in the three dimensions of the space of said acoustic field, said coefficients making it possible to obtain a representation of said acoustic field substantially independent of the characteristics of said acquisition means,
  • - Said encoding filters are also representative of electro-acoustic characteristics of said acquisition means
  • said means for determining encoding filters receive as input at least one of the following parameters: - parameters representative of the positions relative to the center of said means for acquiring all or part of the sensors;
  • said means for determining the parameters received by said means for determining the encoding filters comprising at least one of the following elements:
  • - Fig.1 is a representation of a spherical coordinate system
  • - Fig.2 is a representative diagram of the acquisition means used
  • - Fig.3 is a general flowchart of the method of the invention: - Fig.4 is a flow diagram of the detail of an embodiment of the calibration step of the method of the invention;
  • - Fig.5 is a flow diagram of the detail of an embodiment of the step of determining the encoding filters of the method of the invention
  • - Fig.6 is a detail diagram of an embodiment of the step of applying the encoding filters
  • - Fig.7 is a block diagram of a device suitable for implementing the method of the invention.
  • Figure 1 there is shown a conventional spherical coordinate system, so as to specify the coordinate system to which reference is made in the text.
  • This coordinate system is an orthonormal coordinate system, of O origin and comprising three axes (QX), (OY) and (02).
  • a position denoted x is described by means of its spherical coordinates (r, ⁇ ,), where r denotes the distance from the origin O, ⁇ orientation in the vertical plane and orientation in the horizontal plane.
  • an acoustic field is known if we define at any point at each instant t the acoustic pressure noted p (r, ⁇ , ⁇ , t), whose Fourier transform is noted P (r, ⁇ , ⁇ J) where / designates the frequency.
  • the method of the invention is based on the use of spatiotemporal functions making it possible to describe any sound field in time and in the three dimensions of space.
  • these functions are so-called spherical Fourier-Bessel functions of the first kind, hereinafter called Fourier-Bessel functions.
  • the Fourier-Bessel functions correspond to the solutions of the wave equation and constitute a base which generates all the acoustic fields produced by sources located outside this zone. Any three-dimensional acoustic field can therefore be expressed by a linear combination of the Fourier-Bessel functions, according to the expression of the inverse Fourier-Bessel transform which is expressed:
  • J v (x) is the Bessel function of pre- first species of order v
  • yf ( ⁇ , ⁇ ) is the real spherical harmonic of order / and of term m, with m going from - / to /, defined by:
  • the Fourier-Bessel coefficients are also expressed in the time domain by the coefficients p !> M (t) corresponding to the inverse temporal Fourier transform of the coefficients P ⁇ , m f).
  • the acoustic field is broken down on the basis of functions, where each of the functions is expressed by an optionally infinite linear combination of Fourier-Bessel functions.
  • each elementary sensor is arranged at specific points in space around a predetermined point 4 designated as the center of the acquisition means 1.
  • the position of each elementary sensor can be expressed in space in a spherical coordinate system such as that described with reference to FIG. 1, centered on the center 4 of the acquisition means 1.
  • each sensor 2 n of the acquisition means 1 When exposed to an acoustic field P, each sensor 2 n of the acquisition means 1 delivers a measurement signal c sauwhich corresponds to the measurement made by this sensor in the acoustic field P.
  • the acquisition means 1 deliver a plurality of signals Ci to c / v which are the signals for measuring the acoustic field P by the acquisition means 1.
  • These measurement signals ci to cw delivered by the acquisition means 1 are therefore directly linked to the acquisition capacities of the elementary sensors
  • FIG 3 there is shown a general flowchart of the method of the invention.
  • the method begins with a step 10 for entering parameters and a step 20 for calibrating the acquisition means, which make it possible to define a set of parameters representative of the structural and / or electro-acoustic characteristics of the acquisition means 1.
  • Certain parameters and in particular parameters representative of electro-acoustic characteristics are dependent on the frequency.
  • the input step 10 and the calibration step 20, which is described in more detail with reference to FIG. 4, can be carried out simultaneously or in any order.
  • the method of the invention may include only the input step 10.
  • the steps 10 of input and 20 of calibration make it possible to determine, for one or more sensors, all or part of the following parameters:
  • H n (f) representative of the frequency response of the corresponding sensor 2 n , for each frequency /; to the sensitivity of the sensor 2 n in the direction aJJ);
  • - parameters cf n (f) representative of the power spectral density of the background noise of the sensor 2 ⁇ ;
  • ⁇ (f) specifying a compromise between the fidelity of representation of the acoustic field P and the minimization of the background noise provided by the sensors 2 ⁇ to n and which can take all the values between 0 and 1:
  • all or part of the parameters described is considered to be independent of the frequency.
  • the parameters ⁇ (f), L (f) and ⁇ (k, m k ) ⁇ (f) are representative of the optimization strategies making it possible to control the extraction of spatiotemporal information from the acoustic field P from the measurement signals.
  • ci at C / v and are entered during the entry step 10.
  • the other parameters can be entered during the input step 10 or determined during the calibration step 20.
  • the method of the invention is carried out only with the parameters ⁇ (f), L (f) and the set of parameters x precedeor the set of parameters B n j > m (f) or a combination of parameters x propeland Bn j), so as to have at least one parameter per elementary sensor 2 n .
  • the method comprises a step 30 for determining encoding filters representative of the at least structural and advantageously electro-acoustic characteristics of the acquisition means 1.
  • This step 30, described in more detail with reference to FIG. 5, makes it possible to take into account all the parameters determined during the input steps 10 and / or calibration.
  • These encoding filters are therefore representative at least of the position characteristics of the elementary sensors 2 n relative to the reference point 4 of the acquisition means 1.
  • these filters are also representative of other structural characteristics of the acquisition means 1, such as the orientation of the elementary sensors 2 ⁇ to 2N OR their mutual influences, as well as their electro-acoustic acquisition capacities and in particular their noise of background, their directivity diagram, their frequency response, ...
  • the encoding filters obtained at the end of step 30 can be memorized, so that steps 10, 20 and 30 are only repeated in the event of modification of the acquisition means 1 or of the optimization strategies. .
  • These encoding filters are applied during a step 40 of processing the signals ci to c / v coming from the elementary sensors 2 ⁇ to 2 / v.
  • This processing corresponds to filtering of the signals and to combinations of the filtered signals.
  • this step 40 of processing the measurement signals by applying the encoding filters to these signals a finite number of coefficients representative in time and in the three dimensions of the space of the acoustic field P is issued. These coefficients are so-called Fourier-Bessel coefficients, denoted P ⁇ , m (f) and correspond to a representation of the acoustic field P substantially independent of the characteristics of the acquisition means 1.
  • the calibration step 20 makes it possible to directly determine the coefficients B ni f representative of the acquisition capacities of the acquisition means 1.
  • This step 20 begins with a sub-step 22 of emission of a specific acoustic field towards the acquisition means 1 and with a sub-step 24 of acquisition of measurement signals by the acquisition means 1 exposed to the sound field emitted.
  • step 20 of calibration is implemented using means of generating an acoustic field which comprise only a fixed speaker, assumed to be punctual and with a flat frequency response, the speaker and the acquisition means 1 being placed in an anechoic environment.
  • the loudspeaker emits the same sound field and the acquisition means 1 are placed in the same position but they are oriented in different and known directions.
  • the loudspeaker is in a different position (] ⁇ ⁇ , ⁇ q p , ⁇ q p ) for each field q generated.
  • the acquisition means 1 are thus exposed to an acoustic field q whose Fourier-Bessel coefficients P ⁇ , m , q (f), in the reference frame of the acquisition means 1, are known up to a given order, noted E 3 .
  • the measurement signals delivered following the acquisition sub-step 24 are a finite number of coefficients represented sensitive to the acoustic field q generated, as well as the acquisition capacities of the acquisition means 1.
  • the method includes a modeling sub-step 26 making it possible to determine a representation of the Q acoustic fields emitted during sub-step 22.
  • a modeling matrix P representative of all of the Q known fields to which the acquisition means 1 are successively exposed is determined.
  • This matrix P is a matrix of size (E 3 + l) 2 over Q consisting of the elements P ⁇ , m , q (f), the indices (l, m) designating the line l 2 + l + m and the index q designating column q.
  • the matrix P therefore has the following form:
  • the sound field produced by the loudspeaker is modeled by spherical radiation, thus, in the reference of the acquisition means 1, the coefficients P ⁇ , m , q (f) of each sound field q thus generated are known, thanks to the relation:
  • the coefficients obtained during sub-step 26 are then used during a sub-step 28 in order to determine parameters representative of the structural and / or acoustic characteristics of the acquisition means 1.
  • this sub-step 28 also uses the modeling matrix R determined during sub-step 26.
  • This sub-step 28 begins with the determination of a matrix C representative of all the signals c n ⁇ (t) collected at the output of the N sensors in response to the Q known fields.
  • This matrix is a matrix of size N over Q consisting of the elements C n , q (f), the index n designating the row n and the index q designating the column q.
  • the elements C are deduced from the signals c voyage > q (t) by Fourier transform.
  • the matrix C therefore has the following form:
  • the matrix C is representative of the acquisition capacities of the acquisition means 1 and of the Q acoustic fields emitted.
  • the coefficients B provokej tJtt (f) are determined from the matrices C and P using conventional methods of generalized matrix inversion, applied to the relation which binds C to P.
  • the coefficients R,., /, m (/) are placed in a matrix B determined by the following relation:
  • the matrix B is a matrix of size N on (E 3 + l) 2 consisting of the coefficients B nm if, the index n designating the row n and the indices (l, m) designating the column l 2 + l + m.
  • the matrix B therefore has the following form:
  • the substeps 26 and 28 of the calibration step 20 can be carried out in different ways, depending on the parameters to be determined.
  • the sub-steps 26 and 28 use the propagation times of the waves emitted by the loudspeakers to reach the sensors 2zen.
  • the position of each sensor 2 n is determined using at least three propagation time measurements according to triangulation methods.
  • sub-steps 26 and 28 make it possible to determine, from the signals c n> q (t), the impulse responses of each sensor 2 n when the loudspeaker emits a given pulse.
  • the calibration step 20 allows the determination of the electro-acoustic characteristics of the sensors. It then begins with the determination of the directivity diagram of each sensor 2 n for each frequency / considered, for example, by determining the frequency response of each sensor 2 n for several directions. Then, all or part of the following parameters are determined:
  • This parameter d n (f) can be determined using the usual methods of parameter estimation, for example by applying a method to the least dres squares providing the value of d n (f) which minimizes the error between the real directivity diagram and the modeled directivity diagram.
  • the calibration step 20 also makes it possible to determine the parameter ⁇ A n (J) corresponding to the power spectral density of the background noise of the sensors.
  • the signal delivered by the sensor 2 n is collected in the absence of an acoustic field.
  • the parameter ⁇ A n (f) is determined by means of power spectral density estimation methods, for example the so-called periodogram method.
  • all or part of the sub-steps 22 to 28 is repeated, for example to allow the determination of several types of parameters, certain sub-steps possibly being common to the determination of different types of parameters.
  • the calibration step 20 can also be carried out by means other than those described, such as direct measurements, for example using optical means for measuring the position of each elementary sensor 2 ⁇ relative to the center 4 of the means of acquisition 1.
  • the calibration step 20 can implement a simulation, for example using a computer, of signals representative of the acquisition capacities of the elementary sensors 2 n . It therefore appears that this calibration step 20 makes it possible to determine all or part of the parameters representative of the structural and / or electro-acoustic characteristics of the acquisition means 1, which are used during step 30 of determining the encoding filters. .
  • FIG. 5 a flow diagram of an embodiment of the step 30 for determining the encoding filters is shown.
  • Step 30 includes a sub-step 32 of determining a matrix B representative of the acquisition capacities of the acquisition means 1 or sampling matrix.
  • the matrix B is determined from the parameters ⁇ nj H Mone(f), d n (f), cc ll (f) and B n f) and is a matrix of size N over
  • Certain elements of the matrix R can be directly determined during steps 10 or 20.
  • the matrix B is then supplemented with elements determined from a modeling of the sensors.
  • each sensor n is modeled by a point sensor placed at the position ⁇ n] having a directivity composed of a combination of omnidirectional and bidirectional diagrams of proportion d n (f), oriented in the direction aAf) and having a frequency response H Tar(j).
  • Step 30 then comprises a sub-step 34 for determining an intercorrelation matrix A representative of the resemblance between the signals ci to cw delivered by the sensors 2 ⁇ to 2A / due to the fact that these sensors 2 ⁇ to 2 / v perform measurements on the same sound field P.
  • the matrix A is determined from the sampling matrix B.
  • A is a matrix of size N over N obtained by means of the relation:
  • the matrix A is determined more precisely by using a matrix B completed to an order L 2 according to the method of the previous step.
  • the sub-step 34 of determining the intercorrelation matrix A can be considered as an intermediate calculation step and can therefore be integrated into another sub -step of step 30.
  • Step 30 then comprises a sub-step 36 for determining an encoding matrix E (J) representative of the encoding filters for a given frequency.
  • the matrix E (j) is determined from the matrices A and B and the parameters L (f), ⁇ (f), ⁇ (h, nik) ⁇ (f) e ⁇ admir 2 (f).
  • the matrix E (f) is a matrix of size (L (f) +1) 2 over N consisting of elements E ⁇ > m , n (f), the indices (l, m) designating the line l 2 + l + m and the index n designating the column n.
  • the matrix E (f) therefore has the following form:
  • the matrix E (f) is determined line by line. For each operating frequency / each line E /,nd, of index (l, m) of the matrix E (f) takes the following form:
  • Substeps 32, 34 and 36 for determining the matrices A, B and E (f) are repeated for each frequency / of operation.
  • the parameters are independent of the frequency and the substeps 32, 34 and 36 are carried out only once.
  • Sub-step 36 then directly allows the determination of a matrix E independent of the frequency.
  • parameters FD representative of the encoding filters are determined from the matrix E (f).
  • Each element E ⁇ ⁇ m , n if) of the matrix E (f) represents the frequency response of an encoding filter.
  • Each encoding filter can be described by the FD parameters in different forms.
  • the parameters FD representative of the filters E / , m , " (/) are: - frequency responses, the parameters FD are then directly the E ⁇ , m> f) calculated for certain frequencies /;
  • step 30 of determining the encoding filters delivers parameters FD describing encoding filters representative of the at least structural and / or electro-acoustic capacities of the acquisition means 1.
  • these filters are representative of the following characteristics:
  • step 40 the coefficients p i ⁇ m ( ⁇ representative of the acoustic field P are deduced from the signals Ci to CN coming from the elementary sensors 2 ⁇ to 2A /, by the application of the encoding filters for response in French accordingly E ,, m, .C0 as follows.:
  • coefficients p hm are a finite number of coefficients representative in time and in the three dimensions of the space of the acoustic field and constitute a faithful representation of this acoustic field.
  • other filtering by E7, w , “(/) can be carried out according to different filtering methods, such as for example:
  • the filtering is carried out by means of filtering methods in the frequency domain such as, for example, convolution techniques by blocks;
  • the invention makes it possible to faithfully represent an acoustic field by a substantially independent representation of the characteristics of the acquisition means in the form of Fourier-Bessel coefficients.
  • the method of the invention can be implemented using only the knowledge of the parameters x announcedrepresentative of the position of the sensors 2 n relative to the center 4 of the acquisition means 1 and of the parameters ⁇ and L relating to the optimization strategy. Furthermore, in this simplified embodiment, it is considered that the parameters are independent of the frequency.
  • the matrices A and B are calculated simultaneously or sequentially in any order during sub-steps 32 and 34.
  • the elements B n ⁇ m (f) of the matrix B are then organized as follows: 22
  • the matrix A is obtained from the matrix B by means of the relation:
  • the elements A nn (j) of the matrix A are determined with better precision by the relation:
  • E 2 is the order in which the determination of the matrix ⁇ is carried out and is an integer greater than L.
  • the encoding matrix E representative of the encoding filters is determined from the matrices A and B and from the parameter ⁇ according to the expression:
  • Sub-steps 32, 34 and 36 for determining the matrices A and B then E are repeated for all of the frequencies / of operation.
  • Each element E., m , .. (H corresponds to an encoding filter which integrates the spatial distribution of the sensors 2 n as well as the optimization strategy.
  • phase 40 the signals to CM from sensors 2- ⁇ to 2N are filtered using the encoding filters described by the parameters FD.
  • Each coefficient p l> m (t) delivered is deduced from the signals c- to c ⁇ by applying the filters as follows:
  • the coefficients p lm ( ⁇ are determined by means of filtering methods in the frequency domain, such as for example block convolution techniques.
  • the representation of the sound field therefore takes into account the position of the sensors and the optimization parameters chosen and constitutes a faithful estimate of the sound field.
  • FIG 7 there is shown a block diagram of a device suitable for implementing the method of the invention.
  • a device 50 for representing the acoustic field P is connected to the acquisition means 1 as described with reference to FIG. 2.
  • the device 50 or encoding device is also connected at input to means 60 for determining parameters representative of the structural and / or electro-acoustic characteristics of the acquisition means 1.
  • means 60 include in particular means 62 for entering parameters and calibration means 64 which are suitable for implementing steps 10 and 20 respectively of the method of the invention as described above.
  • the encoding device 50 receives, means 60 for determining the parameters, a plurality of parameters representative of the characteristics of the acquisition means 1 distributed between a signal CL for defining the structural characteristics and a signal CP for configuring the structural characteristics and / or electro-acoustic.
  • the device also receives parameters relating to the representation strategies in a signal OS for optimizing the representation.
  • ⁇ (j) specifying the compromise between the fidelity of representation of the acoustic field and the minimization of the background noise provided by the sensors
  • - a parameter L (f) specifying the order in which the representation is carried out
  • this device 50 includes means 51 for shaping the input signals adapted to deliver from the signals Ci to CN, a signal SI put into corresponding shape.
  • the means 51 comprise analog-to-digital converters, amplifiers or even filtering systems.
  • the device 50 further comprises means 52 for determining the encoding filters which comprise a module 55 for calculating the sampling matrix B, a module 56 for calculating the intercorrelation matrix A, which are both connected to a module 57 for calculating the encoding matrix E (j).
  • This encoding matrix E (f) is used by a module 58 for determining encoding filters which delivers a signal SFD which contains the parameters FD representative of the encoding filters.
  • This SFD signal is used by a processing module 59 which applies the encoding filters to the signal SI in order to deliver a SIFB signal which includes the Fourier-Bessel coefficients representative of the acoustic field P.
  • the device 50 includes a non-volatile memory in which the parameters which constitute the SFD signal which have been determined beforehand are stored.
  • the acquisition means 1 are tested and calibrated by their manufacturer in order to directly supply a memory comprising all the parameters of the SFD signal which it is necessary to integrate into an encoding device in order to carry out the acquisition. of the acoustic field P and to deliver a faithful representation of the latter.
  • this memory only includes the matrices B and possibly A and the device 50 includes means for entering the parameters constituting the optimization signal OS in order to implement the determination of the matrix E (f) d encoding and determination of the FD parameters representative of the encoding filters.
  • the different modules described can be envisaged as required.
EP03749929A 2002-05-07 2003-05-06 Verfahren und system zum repräsentieren eines schallfeldes Expired - Lifetime EP1502475B8 (de)

Applications Claiming Priority (3)

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FR0205741 2002-05-07
FR0205741A FR2839565B1 (fr) 2002-05-07 2002-05-07 Procede et systeme de representation d'un champ acoustique
PCT/FR2003/001410 WO2003096742A1 (fr) 2002-05-07 2003-05-06 Procede et systeme de representation d'un champ acoustique

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EP1502475A1 true EP1502475A1 (de) 2005-02-02
EP1502475B1 EP1502475B1 (de) 2005-07-27
EP1502475B8 EP1502475B8 (de) 2005-09-28

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EP (1) EP1502475B8 (de)
JP (1) JP4293986B2 (de)
KR (1) KR100972419B1 (de)
CN (1) CN1659926B (de)
AT (1) ATE300852T1 (de)
AU (1) AU2003255562B2 (de)
CA (1) CA2484588C (de)
DE (1) DE60301146T2 (de)
DK (1) DK1502475T3 (de)
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WO (1) WO2003096742A1 (de)

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DK1502475T3 (da) 2005-11-28
AU2003255562B2 (en) 2009-04-23
ATE300852T1 (de) 2005-08-15
JP2005531016A (ja) 2005-10-13
CN1659926A (zh) 2005-08-24
CA2484588C (fr) 2013-03-12
US7212956B2 (en) 2007-05-01
KR100972419B1 (ko) 2010-07-27
EP1502475B1 (de) 2005-07-27
CA2484588A1 (fr) 2003-11-20
AU2003255562A1 (en) 2003-11-11
DE60301146T2 (de) 2006-06-01
EP1502475B8 (de) 2005-09-28
US20050177606A1 (en) 2005-08-11
WO2003096742A1 (fr) 2003-11-20
FR2839565B1 (fr) 2004-11-19
KR20050010784A (ko) 2005-01-28
CN1659926B (zh) 2010-05-12
DE60301146D1 (de) 2005-09-01
FR2839565A1 (fr) 2003-11-14
JP4293986B2 (ja) 2009-07-08

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