EP1737271A1 - Réseau de microphones - Google Patents

Réseau de microphones Download PDF

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Publication number
EP1737271A1
EP1737271A1 EP05450112A EP05450112A EP1737271A1 EP 1737271 A1 EP1737271 A1 EP 1737271A1 EP 05450112 A EP05450112 A EP 05450112A EP 05450112 A EP05450112 A EP 05450112A EP 1737271 A1 EP1737271 A1 EP 1737271A1
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Prior art keywords
microphone
microphones
signals
basic
sound field
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EP05450112A
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German (de)
English (en)
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Friedrich Reining
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AKG Acoustics GmbH
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AKG Acoustics GmbH
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Priority to EP05450112A priority Critical patent/EP1737271A1/fr
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    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R5/00—Stereophonic arrangements
    • H04R5/027—Spatial or constructional arrangements of microphones, e.g. in dummy heads
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04S—STEREOPHONIC SYSTEMS 
    • H04S3/00—Systems employing more than two channels, e.g. quadraphonic
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R2201/00—Details of transducers, loudspeakers or microphones covered by H04R1/00 but not provided for in any of its subgroups
    • H04R2201/40—Details of arrangements for obtaining desired directional characteristic by combining a number of identical transducers covered by H04R1/40 but not provided for in any of its subgroups
    • H04R2201/401—2D or 3D arrays of transducers
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04S—STEREOPHONIC SYSTEMS 
    • H04S2400/00—Details of stereophonic systems covered by H04S but not provided for in its groups
    • H04S2400/15—Aspects of sound capture and related signal processing for recording or reproduction
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04S—STEREOPHONIC SYSTEMS 
    • H04S2420/00—Techniques used stereophonic systems covered by H04S but not provided for in its groups
    • H04S2420/11—Application of ambisonics in stereophonic audio systems

Definitions

  • the invention relates to an array microphone, which is formed from at least two basic microphones connected with a signal processor, which make use of an algorithm for processing the signals of the basic microphone.
  • the invention relates to a method for modeling an array microphone made from basic microphones, where the signals of the basic microphones are processed using a filter algorithm.
  • array microphones are used, on the one hand, as natural speech microphones in telephone conversations, for example, in private cars, and, on the other hand, in systems such as navigation systems operated by voice recognition.
  • Adaptive beam formers can, in the process, adapt to movable sources of interference that change over time, for example, during the start phase, the flight phase, and the landing phase of an airplane.
  • An input requirement for the functioning of a beam former is that it must be provided the directions from which useful or interfering sound can be expected. This information, however, can also change over time, or it can be calculated by an additional algorithm with the aid of the microphone signals, for example, if there are several pilots in a cockpit, whose movements are to be tracked.
  • Array microphones essentially consist of an arrangement of individual microphones, which are interconnected by signal technology. In the arrangement of the microphones, a distinction can in principle be made between array microphones in a one-, two- and three-dimensional arrangement. In the case of the one-dimensional arrangement, the microphones are arranged along a line. If microphones that have a spherical directivity pattern (so called omnidirectional microphones) are used, the orientation of the individual microphones is not essential, because they only function as pressure receivers and therefore have an undirected effect in space.
  • the overall directivity pattern, and thus the overall directionality of the array microphone, are the result of a combination of the directivity patterns of the individual microphones with respect to each other, with the use of an algorithm by means of which the microphone signals are processed jointly.
  • the signal-technology connection of the individual microphones can be carried out in the analog or the digital domain.
  • the individual signals of the basic microphones are digitized by means of A/D converters (analog-digital converters) and applied to a signal processing unit.
  • A/D converters analog-digital converters
  • an appropriate algorithm key word "beam forming"
  • the degree of directionality of array microphones can be increased, and interfering sound sources that act from certain directions can be suppressed.
  • a good review of array microphones can be found, for example, in M. Brandstein and D. Wards (Editors), Microphone Arrays, Springer Verlag, 2001 and in the literature cited therein.
  • One component of the algorithm consists of filter coefficient sets, which are characteristic for the arrangement, type, sensitivity, and characteristic of the microphones used, the acoustic environment, and the location of the sound sources.
  • filter coefficient sets it is also possible to take into account different properties of the individual microphones, such as those caused by scattering due to the method of manufacture, aging effects, etc.
  • a filter structure that is frequently used in the literature is known as a "filter and sum beam former" (see, for example, M. Brandstein and D. Wards (Editors), Microphone Arrays, Springer Verlag, 2001, page 159).
  • the individual microphone signals are filtered after the analog-digital conversion using appropriate FIR filters (finite impulse response filters), then added.
  • FIR filters finite impulse response filters
  • the above-mentioned filter coefficient sets are calculated in many applications for a fixed predetermined standard situation and are used as constant parameters in the operation of the array microphone.
  • the individual microphones which comprise the array microphone, are known to specialists under the term "basic microphones", an expression that will also be used in the following description.
  • array microphones function on the principle of constructive or destructive superposition of signals.
  • pressure microphone omnidirectional microphone capsules
  • other pressure gradient microphones microphone capsules whose directivity pattern differs from the omnidirectional characteristic, such as the cardioid pattern
  • the pre-established boundary conditions such as the main direction (this is the direction of maximum sensitivity for the microphone) and/or the direction of rejection (this is the direction of lowest sensitivity of the microphone, where it exhibits the greatest sound suppression)
  • the filter coefficients using optimization algorithms. If the microphone signals are filtered using these filter coefficients, then the given boundary conditions for the entire microphone can be satisfied, more or less.
  • the pressure gradient microphone as basic microphones of the microphone array can bring about a partial improvement during the suppression of interfering sound of the overall microphone.
  • one uses varyingly complex adaptive systems which use the time dimension in addition to the 3 spatial dimensions.
  • the spectral properties of the useful sound or of the interfering sound can only be addressed unsatisfactorily.
  • the measures taken to change or adapt the desired overall microphone properties to fit the given requirements, or to movable sources of useful and/or interfering sound concentrate exclusively on the algorithms in the filter systems.
  • the resulting drawbacks are obvious: insufficient suppression of interfering sound, at times an unacceptable S/N ratio, and a limited degree of directionality (for the selective acquisition of the useful signal), as well as the absence of more efficient measures that would allow one to take into account the frequency dependence of the individual directivity patterns.
  • the objects of the invention are to solve this problem and to provide a method by means of which the signal/noise ratio is clearly improved, reaching a high degree of directionality, which allows flexible control of the microphone properties such as the directivity pattern, with the frequency dependence of the basic microphones being taken into account.
  • the method according to the invention for modeling an array microphone formed from basic microphones, with which the signals of the basic microphones are processed with a filter algorithm is characterized in that the basic microphones are sound field microphones and in that, from the B format signals of an individual sound field microphone, at least one signal is generated, whose directivity pattern is adjusted as a function of the position and type of one or more sound sources to be received, and optionally of sources of interfering sound, and in that the signals of the individual sound field microphones are processed, with the filter algorithm, into a microphone signal.
  • Sound field microphones (sometimes also referred to as B format microphones) consist of several pressure-gradient capsules, arranged on an imaginary spherical surface, which are simply called “capsules” below, in as symmetric as possible an arrangement.
  • B format microphones Sound field microphones
  • the great advantage of sound field microphones is that it is possible to change the directivity pattern of the overall microphone by an appropriate conversion by calculation of the individual signals.
  • B format signals it is possible, using different combinations of the so-called B format signals, to simultaneously generate several different signals.
  • a signal can be directed toward a speaking person, while another signal is directed toward a source of interfering sound.
  • Figure 5 is an outline of the method for the determination of the filter coefficients for a filter algorithm of an array microphone according to the state of the art.
  • the directivity pattern of the individual basic microphones 1, 2, 3, and 4 is unchangeable, with each basic microphone generating only a single signal.
  • the beam forming, which is based on the signals of the basic microphones, is carried out in the subsequent filter 17, which generates the output signal 18 (overall microphone signal).
  • the expected overall characteristic of the array microphone is calculated (step 20).
  • the necessary filter coefficients are then calculated (step 21). This can also occur, for example, in an adaptive process involving several iterative steps, which can optimize the overall properties of the array microphone.
  • the present invention expands the state of the art by an additional dimension.
  • This dimension (in addition to the dimensions of space and time), describes the dynamic changeability of the direction effect of each individual basic microphone, for example, starting from a spherical directivity pattern and ranging to cardioid or a figure-eight directivity pattern.
  • Sound field microphones consist of pressure-gradient capsules, which are arranged as symmetrically as possible in the space formed by the surface of an imaginary sphere, and, in the case of four capsules, on the surface of a virtual tetrahedron.
  • the sound microphone 5 consists ⁇ as shown in Figure 2 ⁇ of four pressure-gradient capsules 10, 11, 12, and 13, where the individual capsules are in a tetrahedral arrangement so that the membranes of the individual capsules are essentially parallel to the tetrahedral surfaces.
  • the individual capsules are thus spherical or arranged around a sphere.
  • Each one of these individual capsules delivers its own signal A, B, C, or D.
  • the symmetrical axis of the directivity pattern of each individual microphone is perpendicular to the membrane or to the corresponding surface of the tetrahedron. Thus, the individual microphones present maxima of their directivity patterns in different directions.
  • the great advantage of the sound field microphones is that, after the storage in memory of the sound events received by the individual capsules, the directivity pattern of the overall microphone can be changed by an appropriate calculation of the individual signals, so it can be changed during the sound reproduction or the subsequent preparation of a sound carrier medium in the desired manner.
  • the sound field can be described using a sound field microphone in a point in space by the spherical harmonics of 0th, 1st, 2nd, ... etc., order, depending on the number and the arrangement of the capsules.
  • the four signals of the individual capsules (the so-called A format) are converted to the so-called B format (W, X, Y, Z), which is conventional in the state of the art.
  • the resulting signals consist of a sphere (W) and three figure-eights (X, Y, Z), which are orthogonal to each other and extend along the three spatial directions.
  • Figure 8 shows the directivity patterns of the separate individual capsule signals.
  • the main directions of the figures-eights are normal with respect to the faces of a cube that circumscribes the tetrahedron ( Figure 9).
  • the sound field recorded at the location of the microphone is split by means of such a microphone and by the associated calculation instruction into the spherical harmonics of 0th and first order.
  • W is a spherical harmonic function of 0th order
  • X, Y, and Z are spherical harmonic functions of 1 st order.
  • Figure 3 shows a block diagram, which schematically illustrates how the signals A, B, C, and D of capsules 10, 11, 12, and 13 of a sound field microphone 5 are converted by a matrix 6 according to the above indicated calculation instruction, into the B format (W, X, Y, and Z). Appropriate amplifiers are switched between the capsules and the matrix. Filters 14, 15, 16, and 17 ensure the equalization of the B format signals.
  • the equalized signals are denoted as W', X', Y', and Z'.
  • any desired directivity pattern with the desired preferential direction for the generated overall signal can be obtained.
  • Such a combination of the individual signals via the B format is also called “synthesizing” or “modeling” a microphone. The weighting can be carried out separately for each signal, which also allows the synthesis of frequency-dependent directivity patterns.
  • a great advantage of such microphones is that the desired directivity patterns can also be adjusted even after the sound event has taken place by an appropriate mixing of the individual B format signals.
  • An additional important advantage of sound field microphones is that, in contrast to pressure-gradient capsules, several signals can be generated simultaneously. These signals, which originate from a single sound-field microphone, can naturally be different, and are produced by different combinations of B format signals. Consequently, it is possible to use a single sound-field microphone to simultaneously acquire sound sources that are arranged in different positions. For example, the directivity pattern of a signal can be directed toward the driver of a private car, while the directivity pattern of another signal is directed toward the front seat passenger.
  • the invention is naturally not limited to sound field microphones of first order.
  • the sound field can also be represented by spherical harmonics of the second order or higher.
  • the sensing of a sound field by a capsule group with overlapping directivity patterns offers a "mathematically clean" possibility to receive and reproduce a spatial sound field (periphony, ambisonics, orthophony).
  • a spatial sound field periphony, ambisonics, orthophony.
  • All the B format signals are orthogonal to each other.
  • the sound field is thus split, by sound field microphones, into mutually orthogonal components.
  • This orthogonality allows a differentiated representation of a sound field, which allows the combination of two or more, optionally weighted, B format signals in a controlled manner to generate a microphone signal having the desired directivity pattern.
  • Separation of the sound field into B format signals, which additionally also contain spherical harmonics of the second order, allows an even more differentiated representation of the sound field and an even higher spatial resolution.
  • a sound field microphone 5' which represents the spherical harmonics up to the second order, requires twelve individual gradient microphone capsules that ⁇ as shown in Figure 4 ⁇ are arranged in the form of a dodecahedron, where each front face carries a capsule. The designation of the capsule starts on the front face with "a” and it ends on the right with "1".
  • a Cartesian coordinate system is assumed, in which the normal vectors of the individual capsules are defined as follows.
  • the B format with the known signals of the 0th order and 1st order W, X, Y, Z must now be enlarged with additional signals corresponding to the spherical signal components of second order. These 5 signals are denoted with the letters R, S, T, U, and V.
  • R, S, T, U, and V the connections between the capsule signals s1, s1 Vietnameses12 and the associated B format signals W, X, Y, Z, R, S, T, U, and V are represented.
  • the array microphone 30 is formed from basic microphones, which here are sound field microphones 5a, 5b, 5c, 5d and thus have several capsules arranged around the imaginary spherical surface.
  • the sound field microphones are essentially coincident microphones, because they are all in an interconnected arrangement. Deviations from these coincident conditions occur only if the capsules cannot be arranged at a certain point because of the space they take up. Nevertheless, the assumption that a microphone is coincident is valid up to a certain frequency.
  • independent filters 22c, 22b, 2c, 22d also called basic microphone filters
  • one or more signals is (are) synthesized first for each sound field microphone on its B format signals.
  • the directivity pattern of each individual signal is adjusted as a function of the given requirements 25, such as the main sound direction(s), direction of the interfering sound, and degree of directionality, frequency response, etc. (step 26).
  • This parameterization is carried out separately for each one of the individual basic microphones and separately for the individual resulting signals as a function of the position and type of the sound sources to be received, and optionally as a function of the interfering sound sources.
  • the array microphone according to the invention has the capacity of splitting up complex requirements ⁇ such as several useful sound directions or interfering sound directions ⁇ into simple partial requirements. These partial requirements are satisfied by several signals, which differ from each other and which are generated in each case by the appropriate combination of B format signals. For each individual partial requirement, an optimal basic microphone filter must accordingly be calculated (on this topic, see the embodiment example below). The result of this procedure produces several signals, all directed toward different sound sources.
  • the synthesis of signals is thus achieved by the linear combination of B format signals.
  • the requirements for how these B format signals should be combined can differ for different frequencies or frequency ranges, on the one hand, to take into account the frequency-dependent directivity patterns of the individual capsules and, on the other hand, to be able to optimally adapt to the frequency characteristic of useful and interfering sound sources.
  • These input requirements (as shown in the embodiment example below, they involve the use of weighting factors, angle and direction data, etc.) correspond to filter coefficients, which are calculated in step 27, by means of which the signals of the basic microphones are synthesized in the basic microphone filters 22a-22d.
  • each basic microphone signal can provide algorithms (one for each basic microphone signal) that optimize the directivity pattern and its frequency variation in accordance with the input requirements.
  • each one of these algorithms in no way takes account of the respective other basic microphones.
  • the signals generated by the basic microphone filter reach the array filter 23, which, taking into account the same input requirements and optionally additional parameters, for example, frequency spectrum of the useful and interfering sound, movable or changing sound sources, spatial expansion of a sound source, etc., generates the starting signal 24 of the array microphone according to the invention.
  • the array filter is essentially a signal processor, which applies a filter algorithm to the individual signals algorithm and generates an output signal from them.
  • an optimization of its filter coefficients can be carried out in accordance with the given input requirements. Via back coupling and iteration, the optimization process can simultaneously optimize the parameters of the individual basic microphone filters.
  • Figure 6 shows the course of the optimization of an array microphone 30 according to the invention.
  • the main direction and the optimal direction effect are set, for example, hypo cardioid, cardioid, hypercardioid, etc., for the individual signals of the basic microphones (step 26).
  • the number of signals generated from the B format of a single basic microphone preferably corresponds to the number of sound sources to be received, including the interfering sound sources to be taken into account. In the case of a single sound source, there may be only one signal for each sound field microphone.
  • any number of signals can be used, depending on the required number.
  • all this involves signal-technology processing by an appropriate number, weighting, and combination of individual B format signals of a basic microphone.
  • the position and type of interfering sound sources constitute an important criterion in this adjustment.
  • the goal is to achieve as good as possible a suppression of interfering sound. This procedure is preferably carried out for all frequencies or several frequency ranges.
  • the information concerning main directions and the direction effect of the individual signal is stored in the form of filter coefficients (which are calculated in step 27) and in the form of an appropriate algorithm in the basic microphone filters 22a-22b.
  • the characteristic of the entire array microphone is calculated (step 28).
  • interfering secondary lobes are determined and compared with the input requirements.
  • modified basic microphone parameters can now be calculated (step 29).
  • the basic microphone parameters are used again as input for the calculation of the overall characteristic, until it satisfies the input requirements or approximates such.
  • modified filter coefficients can be calculated for the array filter 23, in a calculation that is performed while taking into consideration the basic microphone parameters (step 32 in Figure 6).
  • An array microphone is considered, which is formed from four sound field microphones.
  • the input requirement one uses a known direction from which the useful sound is expected. Interfering sound from all other directions should be suppressed to the extent possible. However, the type and manner of the interfering sound varies.
  • All the sound field microphones are adjusted to hypercardioid, to achieve a maximum degree of directionality, and are directed with their main direction in each case being toward the useful sound source.
  • one synthesizes--for example, for one of the sound field microphones--one additional basic microphone signal which, however, presents a temporally varying directivity pattern, that is, one that , for example, rotates in space with an angular speed to allow the dynamic acquisition of interfering sound sources from all directions (scanning).
  • each basic microphone signal can be optimized in order to suppress the interfering sound source to the extent possible (for example, by rotating the basic microphone main direction, or changing the directivity pattern). This optimization can be carried out for different frequencies or frequency ranges, for example, to eliminate an interfering sound source that presents only high frequencies. The interfering sound source is thus optimally eliminated for each individual basic microphone.
  • the filter algorithm of the filter 23 can also take into account this information in the calculation of its filter parameters and thus optimize the output signal 24.
  • the filter algorithm can be a least mean square (LMS) algorithm that, with the help of Lagrange multipliers, also takes into account that information via the basic microphone signals in the form of secondary conditions.
  • LMS least mean square
  • an appropriate measuring means an additional algorithm, which, for example, can make distinctions based on the spectral distribution between useful and interfering sound
  • the entire process can be run several times using an iteration procedure.
  • Figure 7 shows an embodiment of the invention.
  • the starting point is the acquisition of two persons (driver and front seat passenger in a private car) by four basic microphones and a filter algorithm or a so-called beam former algorithm.
  • Conventional procedures calculate, via a filter algorithm, a double lobe, as shown in the top part of Figure 7.
  • omnidirectional capsules are used as basic microphones, the useful signal or interfering signal directions are not adjusted better, however, the suppression of the interfering sound is subject only to the filter algorithm or the beam former algorithm.
  • pressure gradient capsules they can optimally be directed only toward one useful sound source (or suboptimally to both). The suppression of interfering sound can now also occur at the base capsules, but only for a fixed or predetermined position of the useful sound sources.
  • a set of basic microphone signals is first synthesized, where the signals are oriented in the direction toward the driver; an additional set of basic microphone signals is provided, which are all rotated in the direction of the front seat passenger.
  • the interfering sound signals are already largely eliminated by the optimal orientation of the basic microphones 5a-5b or the basic microphone signals for each useful sound direction.
  • a one-dimensional array with 4 sound field microphones arranged equidistantly is intended to be directed toward the driver and the front seat passenger in a private car in an optimal manner and to optimally dampen the noise from the air cooler of the dashboard (in the middle close to the stick shift).
  • a FIR filter--in filter 23 an LMS (least mean squares algorithm)--is to be used.
  • the directions of the useful sound (driver, front seat passenger) as well as of the interfering sound (ventilator, not shown) are assumed to be constant.
  • the middle of the array (between the two lower sound-field microphones) is taken as the origin of the underlying overall coordinate system.
  • the XZ plane is defined as the plane between the driver and the front seat passenger.
  • the array thus extends one dimensionally along the Y axis.
  • the driver and front seat passenger forms, with the X axis, an angle of ⁇ 30° and is located in the XY plane; the ventilator (not shown) is in the direction of the negative Z axis below the array.
  • the individual B format signals are generated with the application of the above-mentioned transformation formulas.
  • All four sound field microphones are oriented in space in such a manner that the figure-eight signals of the B format assumes a position, with respect to the middle of each sound field microphone, along the X direction (X' is used to denote the figure-eight lobe of the first sound field microphone, which is parallel to the X axis of the overall coordinate system), the Y direction, and the Z direction.
  • the figure-eight lobes of the 3 additional sound field microphones are denoted X", Y", Z", X"', etc.
  • hypercardioid patterns are generated from the B format signals. This occurs with the application of corresponding weighting factors between the omnidirectional signal and figure-eight signals.
  • the weighting factor for the omnidirectional signal W is 0.25 and for the figure-eight signal X, Y, or Z or linear combinations thereof, it is 0.75.
  • Basic microphone signal directed to the driver 0.25 ⁇ W + 0.75 ⁇ X ⁇ cos 30 ⁇ Y ⁇ sin 30
  • Basic microphone signal directed to the front seat passenger 0.25 ⁇ W + 0.75 ⁇ X ⁇ cos 30 + Y ⁇ sin 30
  • Basic microphone signal directed to the ventilator 0.25 ⁇ W ⁇ 0.75 ⁇ Z .
  • all 4 sound field microphones are parameterized to the same angle of 30°, although the value of the angle deviates slightly from 30° for each microphone.
  • weighting factors (0.25 or 0.75) for the B format signals and angle indications (30°) for directions from which sound is expected, the individual sound sources are selectively acquired even before the processing of the individual basic microphone signals in the filter 23.
  • the beam forming algorithm which is here considered as an embodiment example, works on the principle of energy minimization.
  • the sound source for example, human mouth
  • the sound source sends out a wave front, which is to be received by the spatial arrangement of of each microphone at a different time.
  • the calculation is carried out in a small band, so the result can be interpreted as a vector of complex frequency points. If one runs the calculation in a loop, with the frequency as the loop parameter, one obtains a complex frequency response from which, as realization, a FIR filter can be determined.
  • the goal of the following representation is to force a minimization of the energy in the output signal, with the exception of the desired main direction, for which the energy should be 1.
  • w n is the complex weight, the nth entry in the optimized vector, whose magnitude and phase have an effect on the microphone signal;
  • H n ( ⁇ i , ⁇ ) is the complex directivity pattern for the individual frequency ⁇ i .
  • the superscript H in formula A5 indicates that the vector w is a Hermitian vector w, which is characterized in that w* (that is, the conjugate complex vector of w) is equal to the vector w.
  • ⁇ 0 is the instantaneously desired main acoustic incidence direction.
  • w ⁇ ⁇ A ⁇ 1 ⁇ m ⁇ 0
  • the matrix A represents the cross correlation between the individual microphone signals.
  • the autocorrelation functions of the individual microcapsules are on the main diagonals.
  • the so-called filter and sum beam former can be considered a special case of a beam forming algorithm calculated by the above method, when the filters are exclusively of the all-pass type (in the literature, also called delay and sum beam former). This type can be easily calculated using the above calculation procedure by omitting all the components of the matrix A, which cannot be found in the main diagonal.
  • the calculated complex weights w in the above-mentioned loop, yield a complex frequency response, which can be transformed into the time domain by inverse Fourier transformation (IFT), yielding the coefficients of a FIR (finite impulse response) filter.
  • IFT inverse Fourier transformation
  • Each basic microphone signal which is directed to the same sound source, is filtered using the specifically calculated FIR filter, then added.
  • This procedure is also called beam forming.
  • the input requirement (3 sound sources) was shaped into 3 beams, so 3 beam-forming algorithms are calculated in parallel and are available at the end.
  • the two useful signals produced in this manner are added to form a single useful signal, the interfering signal of the ventilator can now, after appropriate damping, also be subtracted from the useful signal, which can result in an additional improvement of the S/N ratio.
  • the array microphone consists of sound field microphones in an arrangement that is not equidistant. Because the principle of the beam former results from the superposition of waves having different phases, one can easily see that for wavelengths in the range of the capsule separation, constructive or destructive interference is very pronounced. Then, to cover as broad a frequency range as possible with the beam former algorithm, the capsule separations are chosen to be as different as possible.
  • the invention is not limited to the represented embodiment example; rather, it can be modified in different manners.
  • four basic microphones is not compulsory.
  • the arrangement of the basic microphones can also be in two or three dimensions. Sound field microphones of the 3rd or higher order would also be conceivable. It is also possible to combine sound field microphones with different capsule numbers and arrangements together in one array.
  • the noncoincident arrangement (array) of coincident (or at least approximately coincident) microphones, which are interconnected by signal technology, is important.
  • the different steps do not necessarily have to be in the described form, the essential factor being that one or more signals is/are first synthesized from the B format signals of each individual sound field microphone, which in part already satisfy the input requirements or partial input requirements, with these signals then being processed with a signal processor using a filter algorithm to form the overall microphone signal.
EP05450112A 2005-06-23 2005-06-23 Réseau de microphones Withdrawn EP1737271A1 (fr)

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Cited By (15)

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WO2009062211A1 (fr) * 2007-11-13 2009-05-22 Akg Acoustics Gmbh Détermination de position de sources sonores
WO2009062213A1 (fr) * 2007-11-13 2009-05-22 Akg Acoustics Gmbh Microphone ayant deux transducteurs de gradient de pression
WO2009062214A1 (fr) * 2007-11-13 2009-05-22 Akg Acoustics Gmbh Procédé de synthétisation d'un signal de microphone
WO2009062212A1 (fr) * 2007-11-13 2009-05-22 Akg Acoustics Gmbh Microphone comprenant trois transducteurs de gradient de pression
WO2009062210A1 (fr) * 2007-11-13 2009-05-22 Akg Acoustics Gmbh Microphone
US8345898B2 (en) 2008-02-26 2013-01-01 Akg Acoustics Gmbh Transducer assembly
EP2592846A1 (fr) * 2011-11-11 2013-05-15 Thomson Licensing Procédé et appareil pour traiter des signaux d'un réseau de microphones sphériques sur une sphère rigide utilisée pour générer une représentation d'ambiophonie du champ sonore
KR20140091578A (ko) * 2011-11-11 2014-07-21 톰슨 라이센싱 음장의 앰비소닉스 표현을 생성하기 위해 사용되는 강체구상에서의 구면 마이크로폰 배열의 신호들을 처리하기 위한 방법 및 장치
US9173048B2 (en) 2011-08-23 2015-10-27 Dolby Laboratories Licensing Corporation Method and system for generating a matrix-encoded two-channel audio signal
US9407869B2 (en) 2012-10-18 2016-08-02 Dolby Laboratories Licensing Corporation Systems and methods for initiating conferences using external devices
WO2017152601A1 (fr) * 2016-03-10 2017-09-14 中兴通讯股份有限公司 Procédé et terminal de détermination de microphone
CN108632711A (zh) * 2018-06-11 2018-10-09 广州大学 扩声系统增益自适应控制方法
WO2019174725A1 (fr) * 2018-03-14 2019-09-19 Huawei Technologies Co., Ltd. Dispositif et procédé de codage audio
US10477310B2 (en) 2017-08-24 2019-11-12 Qualcomm Incorporated Ambisonic signal generation for microphone arrays
CN110579275A (zh) * 2019-10-21 2019-12-17 南京南大电子智慧型服务机器人研究院有限公司 一种基于球形矢量传声器阵列实现声场分离的方法

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CN101911721B (zh) * 2007-11-13 2014-04-23 Akg声学有限公司 合成麦克风信号的方法
WO2009062213A1 (fr) * 2007-11-13 2009-05-22 Akg Acoustics Gmbh Microphone ayant deux transducteurs de gradient de pression
WO2009062214A1 (fr) * 2007-11-13 2009-05-22 Akg Acoustics Gmbh Procédé de synthétisation d'un signal de microphone
WO2009062212A1 (fr) * 2007-11-13 2009-05-22 Akg Acoustics Gmbh Microphone comprenant trois transducteurs de gradient de pression
WO2009062210A1 (fr) * 2007-11-13 2009-05-22 Akg Acoustics Gmbh Microphone
CN101911721A (zh) * 2007-11-13 2010-12-08 Akg声学有限公司 合成麦克风信号的方法
EP2262277A1 (fr) * 2007-11-13 2010-12-15 AKG Acoustics GmbH Ensemble de microphones
WO2009062211A1 (fr) * 2007-11-13 2009-05-22 Akg Acoustics Gmbh Détermination de position de sources sonores
US8472639B2 (en) 2007-11-13 2013-06-25 Akg Acoustics Gmbh Microphone arrangement having more than one pressure gradient transducer
US8345898B2 (en) 2008-02-26 2013-01-01 Akg Acoustics Gmbh Transducer assembly
US9173048B2 (en) 2011-08-23 2015-10-27 Dolby Laboratories Licensing Corporation Method and system for generating a matrix-encoded two-channel audio signal
CN104041074A (zh) * 2011-11-11 2014-09-10 汤姆逊许可公司 处理用于产生声场的高保真度立体声响复制表示的刚性球上的球形麦克风阵列的信号的方法和装置
CN104041074B (zh) * 2011-11-11 2017-04-12 杜比国际公司 处理用于产生声场的高保真度立体声响复制表示的刚性球上的球形麦克风阵列的信号的方法和装置
KR20140091578A (ko) * 2011-11-11 2014-07-21 톰슨 라이센싱 음장의 앰비소닉스 표현을 생성하기 위해 사용되는 강체구상에서의 구면 마이크로폰 배열의 신호들을 처리하기 위한 방법 및 장치
WO2013068284A1 (fr) * 2011-11-11 2013-05-16 Thomson Licensing Procédé et appareil de traitement de signaux d'un réseau de microphones sphérique sur une sphère rigide utilisé pour générer une représentation d'ambiophonie du champ sonore
EP2592846A1 (fr) * 2011-11-11 2013-05-15 Thomson Licensing Procédé et appareil pour traiter des signaux d'un réseau de microphones sphériques sur une sphère rigide utilisée pour générer une représentation d'ambiophonie du champ sonore
KR20140089601A (ko) * 2011-11-11 2014-07-15 톰슨 라이센싱 사운드 필드의 앰비소닉스 표현을 생성하는데 사용되는 강체구 상의 구면 마이크로폰 어레이의 신호들을 프로세싱하는 방법 및 장치
US9420372B2 (en) 2011-11-11 2016-08-16 Dolby Laboratories Licensing Corporation Method and apparatus for processing signals of a spherical microphone array on a rigid sphere used for generating an ambisonics representation of the sound field
US9407869B2 (en) 2012-10-18 2016-08-02 Dolby Laboratories Licensing Corporation Systems and methods for initiating conferences using external devices
WO2017152601A1 (fr) * 2016-03-10 2017-09-14 中兴通讯股份有限公司 Procédé et terminal de détermination de microphone
US10477310B2 (en) 2017-08-24 2019-11-12 Qualcomm Incorporated Ambisonic signal generation for microphone arrays
CN111819862A (zh) * 2018-03-14 2020-10-23 华为技术有限公司 音频编码设备和方法
WO2019174725A1 (fr) * 2018-03-14 2019-09-19 Huawei Technologies Co., Ltd. Dispositif et procédé de codage audio
US11632626B2 (en) 2018-03-14 2023-04-18 Huawei Technologies Co., Ltd. Audio encoding device and method
CN111819862B (zh) * 2018-03-14 2021-10-22 华为技术有限公司 音频编码设备和方法
CN108632711A (zh) * 2018-06-11 2018-10-09 广州大学 扩声系统增益自适应控制方法
CN108632711B (zh) * 2018-06-11 2020-09-04 广州大学 扩声系统增益自适应控制方法
CN110579275A (zh) * 2019-10-21 2019-12-17 南京南大电子智慧型服务机器人研究院有限公司 一种基于球形矢量传声器阵列实现声场分离的方法

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