Classifications

• • G—PHYSICS
  • G10—MUSICAL INSTRUMENTS; ACOUSTICS
  • G10H—ELECTROPHONIC MUSICAL INSTRUMENTS; INSTRUMENTS IN WHICH THE TONES ARE GENERATED BY ELECTROMECHANICAL MEANS OR ELECTRONIC GENERATORS, OR IN WHICH THE TONES ARE SYNTHESISED FROM A DATA STORE
  • G10H1/00—Details of electrophonic musical instruments
  • G10H1/0091—Means for obtaining special acoustic effects
• • G—PHYSICS
  • G10—MUSICAL INSTRUMENTS; ACOUSTICS
  • G10H—ELECTROPHONIC MUSICAL INSTRUMENTS; INSTRUMENTS IN WHICH THE TONES ARE GENERATED BY ELECTROMECHANICAL MEANS OR ELECTRONIC GENERATORS, OR IN WHICH THE TONES ARE SYNTHESISED FROM A DATA STORE
  • G10H1/00—Details of electrophonic musical instruments
• • 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 present invention relates to the processing of sound data.
• Techniques relating to the propagation of a sound wave in three-dimensional space involving in particular a simulation and / or a specialized sound reproduction, implement audio signal processing methods applied to the simulation of acoustic and psycho-acoustic phenomena .
• Such processing methods provide for spatial encoding of the acoustic field, its transmission and its spatial reproduction on a set of loudspeakers or on headphones of a stereophonic headset.
• a first category of treatments relates to processes for synthesizing room effects, or more generally environmental effects. From a description of one or more sound sources (signal emitted, position, orientation, directivity, or other) and based on a room effect model (involving room geometry, or even an acoustic perception desired), we calculate and describe a set of elementary acoustic phenomena (direct waves, reflected or diffracted), or even a macroscopic acoustic phenomenon (reverberant and diffuse field), allowing to translate the spatial effect at the level of a listener located at a chosen point of auditory perception, in three-dimensional space. We then calculate a set of signals typically associated with reflections
• Secondary sources active by re-emission of a main wave received, having a spatial position attribute
• a late reverberation decorrelated signals for a diffuse field
• a second category of process concerns the positional or directional rendering of sound sources. These methods are applied to signals determined by a method of the first category described above.
• Methods according to this second category make it possible to obtain signals to be broadcast over loudspeakers or headphones, in order to finally give a listener the auditory impression of sound sources placed at respective predetermined positions, around the auditor.
• the methods according to this second category are qualified as "creators of three-dimensional sound images", because of the distribution in three-dimensional space of the feeling of the position of the sources by a listener.
• Methods according to the second category generally include a first step of spatial encoding of elementary acoustic events which produces a representation of the sound field in three-dimensional space. In a second step, this representation is transmitted or stored for deferred use. In a third decoding step, the decoded signals are delivered to speakers or headphones of a playback device.
• the present invention falls rather into the second category mentioned above. It concerns in particular the spatial encoding of sound sources and a specification of the three-dimensional sound representation of these sources. It applies as well to an encoding of "virtual" sound sources (applications where sound sources are simulated such as games, a spatial conference, or other), as an "acoustic" encoding of a natural sound field, when taking sound through one or more three-dimensional networks of microphones.
• Ambisonic encoding which will be described in detail later, consists in representing signals relating to one or more sound waves in a base of spherical harmonics (in spherical coordinates notably implying an elevation angle and an azimuthal angle, characterizing a direction of the sound (s)).
• the components representing these signals and expressed in this base of spherical harmonics are also a function, for waves emitted in the near field, of a distance between the sound source emitting this field and a point corresponding to the origin of the harmonic base. spherical. More specifically, this distance dependence is expressed as a function of the sound frequency, as will be seen below.
• This ambisonic approach offers a large number of possible functionalities, in particular in terms of simulation of virtual sources, and, in general, has the following advantages:
• the representation of acoustic phenomena is scalable: it offers a spatial resolution which can be adapted to different situations. Indeed, this representation can be transmitted and used as a function of rate constraints during the transmission of the encoded signals and / or of limitations of the restitution device;
• the ambisonic representation is flexible and it is possible to simulate a rotation of the sound field, or even, at restitution, to adapt the decoding of ambisonic signals to any restitution device, of various geometries.
• the encoding of virtual sources is essentially directional.
• the encoding functions amount to calculating gains which depend on the incidence of the sound wave expressed by the spherical harmonic functions which depend on the elevation angle and the azimuth angle in spherical coordinates.
• the loudspeakers, at restitution are far away. he this results in a distortion (or curvature) of the shape of the reconstructed wave fronts.
• the components of the sound signal in the base of spherical harmonics, for a near field in fact also depend on the distance from the source and on the sound frequency.
• these components can be expressed mathematically in the form of a polynomial, the variable of which is inversely proportional to the aforementioned distance and to the sound frequency.
• the ambisonic components in the sense of their theoretical expression, are divergent in the low frequencies and, in particular, tend towards infinity when the sound frequency decreases towards zero, when they represent a sound in near field emitted by a source located at a finite distance.
• This mathematical phenomenon is known, in the field of ambisonic representation, already for order 1, by the term "bass boost", in particular by: - .A.GERZON, "General Metatheory of Audi tory
• This phenomenon becomes particularly critical for high spherical harmonic orders involving high power polynomials.
• spatial aliasing a processing artefact that appears because of the distance between the microphones, unless you choose a virtual microphone mesh tight in space, which weighs down the processing;
• this technique is difficult to transpose to a real case of sensors to be placed on a network, in the presence of a real source, at acquisition;
• the three-dimensional sound representation is implicitly subject to a fixed radius of the rendering device because the ambisonic decoding must be done, here, on a network of loudspeakers of the same dimensions as the initial microphone network, this document does not offering no way to adapt encoding or decoding to other sizes of rendering devices.
• this document presents a horizontal network of sensors, which supposes that the acoustic phenomena which are taken into account here, propagate only in horizontal directions, which excludes any other direction of propagation and which, therefore, does not represent the physical reality of an ordinary sound field.
• An object of the present invention is to provide a method for processing, by encoding, transmission and reproduction, any type of sound field, in particular the effect of a sound source in the near field.
• Another object of the present invention is to provide a method allowing the encoding of virtual sources, not only in direction, but also in distance, and to define a decoding adaptable to any rendering device.
• Another object of the present invention is to provide a robust processing method for sounds of all sound frequencies (including low frequencies), especially for taking sound from natural acoustic fields using three-dimensional networks of microphones.
• the present invention provides a process for processing sound data, in which: a) signals representative of at least one sound propagating in three-dimensional space and coming from a source located at a first distance are coded from a reference point, to obtain a representation of the sound by components expressed in a base of spherical harmonics, of origin corresponding to said reference point, and b) a compensation for a near field effect is applied to said components by filtering which is a function of a second distance defining substantially, for a sound reproduction by a reproduction device, a distance between a reproduction point and a hearing perception point.
• said source being far from the reference point
• a filter is applied, the coefficients of which, each applied to a component of order m, are expressed analytically in the form of the inverse of a polynomial of power m, the variable of which is inversely proportional to the sound frequency and to said second distance, to compensate for a near field effect at the level of the restitution device.
• said source being a virtual source provided at said first distance
• the numerator is a polynomial of power m, the variable of which is inversely proportional to the sound frequency and to said first distance, to simulate a field effect close to the virtual source, and
• the denominator is a polynomial of power m, the variable of which is inversely proportional to the sound frequency and to said second distance, to compensate for the effect of the near field of the virtual source in the low sound frequencies.
• the coded and filtered data in steps a) and b) are transmitted to the restitution device with a parameter representative of said second distance.
• the restitution device comprising means for reading a memory medium, the coded and filtered data are stored on a memory medium intended to be read by the restitution device. in steps a) and b) with a parameter representative of said second distance.
• an adaptation filter is applied to the coded and filtered data, the coefficients of which are a function of said seconds and third distances.
• the coefficients of this adaptation filter, each applied to a component of order m are expressed analytically in the form of a fraction, of which:
• the numerator is a polynomial of power m, the variable of which is inversely proportional to the sound frequency and to said second distance,
• the denominator is a polynomial of power m, the variable of which is inversely proportional to the sound frequency and to said third distance.
• step b for the implementation of step b), provision is made: - for components of even order m, digital audio filters in the form of a cascade of cells of order two; and
• digital audio filters in the form of a cascade of cells of order two and an additional cell of order one.
• the coefficients of a digital audio filter, for a component of order m are defined from the numerical values of the roots of said polynomials of power m.
• the above-mentioned polynomials are Bessel polynomials.
• a microphone comprising a network of acoustic transducers arranged substantially on the surface of a sphere whose center corresponds substantially to said reference point, in order to obtain said signals representative of at least one sound. propagating in three-dimensional space.
• a global filter is applied in step b) on the one hand, to compensate for a near field effect as a function of said second distance and, on the other hand, to equalize the signals coming from the transducers to compensate for a directivity weighting of said transducers.
• a number of transducers is provided as a function of a chosen total number of components to represent the sound in said base of spherical harmonics.
• step a) a total number of components is chosen from the base of spherical harmonics to obtain, at the restitution, a region of the space around the point of perception in which the restitution of the sound is faithful and whose dimensions are increasing with the total number of components.
• a reproduction device comprising a number of loudspeakers at least equal to said total number of components.
• a reproduction device comprising at least a first and a second loudspeaker arranged at a chosen distance from a listener, for this listener, information is obtained on the feeling expected from the position in the space of sound sources located at a predetermined reference distance from the listener for the application of a technique called "binaural synthesis” or "transaural", and
• step b) The compensation of step b) is applied with said reference distance substantially as a second distance.
• a reproduction device comprising at least a first and a second loudspeaker arranged at a chosen distance from a listener, - information is obtained for this listener about feeling the position in the space of sound sources located at a predetermined reference distance from the auditor, and
• an adaptation filter is applied to the coded and filtered data in steps a) and b), the coefficients of which are a function of the second distance and substantially of the reference distance.
• the restitution device comprises a helmet with two earphones for the respective ears of the listener
• steps a) and b) are applied for respective signals intended to supply each earpiece, with, as first distance, respectively a distance separating each ear from a position of a source to be restored in the reproduction space.
• a matrix system is formed in steps a) and b) comprising at least:
• the reproduction device upon return the reproduction device comprises a plurality of loudspeakers arranged substantially at the same distance from the point of auditory perception, and
• a matrix system comprising said matrix result of compensated components and a predetermined decoding matrix, specific to the restitution device, and
• the present invention also relates to a sound acquisition device, comprising a microphone provided with a network of acoustic transducers arranged substantially on the surface of a sphere.
• the device further comprises a processing unit arranged for:
• a filtering which is a function, on the one hand, of a distance corresponding to the radius of the sphere and, on the other hand, of a reference distance.
• the filtering carried out by the processing unit consists, on the one hand, in equalizing, as a function of the radius of the sphere, the signals coming from the transducers to compensate for a weighting of directivity of said transducers and, on the other hand, in compensate for a near field effect as a function of said reference distance.
• FIG. 1 schematically illustrates a system of acquisition and creation, by simulation of virtual sources, of sound signals, with encoding, transmission, decoding and restitution by a spatialized restitution device
• FIG. 4 illustrates a representation by a three-dimensional metric in a coordinate system of spherical coordinates, of spherical harmonics Y ⁇ n of different orders;
• - Figure 5 is a diagram of the variations of the module of radial functions j a (kr), which are spherical Bessel functions, for successive values of order m, these radial functions intervening in the ambisonic representation of a field of sound pressure;
• - Figure 6 shows the amplification due to the near field effect for different successive orders m, in particular in the low frequencies;
• FIG. 7 schematically shows a playback device comprising a plurality of speakers
• FIG. 8 shows schematically the parameters involved in the ambisonic encoding, with a directional encoding, as well as a distance encoding according to one invention
• FIG. 11A shows a reconstruction of the near field with compensation, within the meaning of the present invention, for a spherical wave in the horizontal plane;
• FIG. 11B represents the initial wavefront, coming from a source S;
• FIG. 12 schematically represents a filtering module for adapting the received and pre-compensated ambisonic components to the encoding for a distance of reference R as a second distance, to a reproduction device comprising a plurality of loudspeakers arranged at a third distance R 2 from a point of auditory perception;
• - Figure 13A schematically shows the arrangement of a sound source M, during playback, for a listener using a playback device applying binaural synthesis, with a source emitting in the near field;
• - Figure 13B schematically shows the steps of encoding and decoding with near field effect in the context of the binaural synthesis of Figure 13A which is combined encoding / decoding ambisonic;
• - Figure 14 schematically shows the processing of signals from a microphone comprising a plurality of pressure sensors arranged on a sphere, for illustrative purposes, by ambisonic encoding, equalization and near field compensation within the meaning of the invention.
• FIG. 1 shows by way of illustration a global sound spatialization system.
• a module simulating a virtual scene defines a sound object as a virtual source of a signal, for example monophonic, of position chosen in three-dimensional space and which defines a direction of sound. There may also be specifications for the geometry of a virtual room, to simulate reverberation of sound.
• a processing module 11 applies management of one or more of these sources with respect to a listener (definition of a virtual position of the sources with respect to this listener). It implements a room effect processor to simulate reverberations or the like by applying common delays and / or filtering.
• the signals thus constructed are transmitted to a module 2a for spatial encoding of the elementary contributions of the sources.
• a natural sound recording can be carried out as part of a sound recording by one or more.
• microphones arranged in a chosen manner in relation to the real sources (module 1b).
• the signals picked up by the microphones are encoded by a module 2b.
• the acquired and encoded signals can be transformed according to an intermediate representation format (module 3b), before being mixed by the module 3 with the signals generated by the module la and encoded by the module 2a (from virtual sources).
• the mixed signals are then transmitted, or even memorized on a medium, for later playback (arrow TR). They are then applied to a decoding module 5, for playback on a playback device 6 comprising speakers.
• the decoding step 5 can be preceded by a step for manipulating the sound field, for example by rotation, by means of a processing module 4 provided upstream of the decoding module 5.
• the rendering device can be in the form of a multiplicity of loudspeakers, arranged for example on the surface of a sphere in a three-dimensional (peripheric) configuration to ensure, during the rendering, in particular a feeling of a direction sound in three-dimensional space.
• an auditor generally places at the center of the sphere formed by the network of loudspeakers, this center corresponding to the point of auditory perception mentioned above.
• the loudspeakers of the reproduction device can be arranged in a plane (two-dimensional panoramic configuration), the loudspeakers being arranged in particular on a circle and the listener usually placing themselves in the center of this circle.
• the restitution device can be in the form of a "surround" type device (5.1).
• the reproduction device can be in the form of a headset with two headphones for a binaural synthesis of the restored sound, which allows the listener to feel a direction of the sources in three-dimensional space, as will be seen in detail below.
• a reproduction device with two loudspeakers, for feeling in three-dimensional space can also be in the form of a transaural reproduction device, with two loudspeakers arranged at a chosen distance from a listener.
• FIG. 2 We now refer to FIG. 2 to describe a spatial encoding and a decoding for a three-dimensional sound reproduction, of elementary sound sources.
• the signal from a source 1 to N is transmitted to a spatial encoding module 2, as well as its position (real or virtual). Its position can be defined as well in terms of incidence (direction of the source seen by the listener) as in terms of distance between this source and a listener.
• the plurality of signals thus encoded allows to obtain a multi-channel representation of a global sound field.
• the encoded signals are transmitted (arrow TR) to a sound reproduction device 6, for sound reproduction in three-dimensional space, as indicated above with reference to FIG. 1.
• the set of weighting factors B ⁇ n which are implicitly a function of the frequency, thus describe the pressure field in the zone considered. For this reason, these factors are called "spherical harmonic components" and represent a frequency expression of the sound (or of the pressure field) in the base of the spherical harmonics Y n .
• spherical harmonics The angular functions are called “spherical harmonics" and are defined by:
• the spherical harmonics form an orthonormee base where the scalar products between harmonic components and, in general between two functions F and G, are respectively defined by:
• Spherical harmonics are real bounded functions, as shown in Figure 4, as a function of the order m and the indices n and ⁇ .
• the dark and light parts correspond respectively to the positive and negative values of the spherical harmonic functions.
• the radial functions j m (kr) are spherical Bessel functions, the modulus of which is illustrated for some values of the order m in Figure 5.
• the ambisonic representation of the sound is however less satisfactory as one moves away from the origin O. This effect becomes critical in particular for high sound frequencies (of short wavelength) . It is therefore advantageous to obtain a number of ambisonic components which is as large as possible, which makes it possible to create a region of space around the point of perception, in which the reproduction of the sound is faithful and whose dimensions are increasing with the total number of components.
• an ambisonic system takes into account a subset of spherical harmonic components, as described above.
• a system of order M when it takes into account ambisonic components of index m ⁇ M.
• the reproduction device comprises loudspeakers arranged on the surface of a sphere ("periphery"), it is in principle possible to use as many harmonics as there are loudspeakers.
• the pressure signal S designates the pressure signal carried by a plane wave and picked up at point O corresponding to the center of the sphere in FIG. 3 (origin of the base in spherical coordinates).
• the incidence of the wave is described by the azimuth ⁇ and the elevation ⁇ .
• the expression of the components of the field associated with this plane wave is given by the relation:
• the encoding produces signals which differ from the original signal only by a real, finite gain, which corresponds to a purely directional encoding (relation [A3]);
• the additional filter Fjjjf '( ⁇ ) encodes distance information by introducing, in the expression of the ambisonic components, complex amplitude relationships which depend on the frequency, as expressed in the relation [A5].
• this additional filter is of the "integrator" type, with an increasing and diverging amplification effect (unbounded) as the sound frequencies decrease towards zero.
• a reproduction device comprises a plurality of loudspeakers HPi, arranged at the same distance R, in the example described, from an auditory perception point P.
• R distance
• the point M corresponds to the position of a source (real or virtual) located at the first distance p, stated above, from the reference point O.
• a near field pre-compensation is introduced at the very stage of encoding, this compensation involving filters of the form
• coefficients of this compensation filter are increasing with the frequency of the sound and, in particular, tending towards zero, for the low frequencies.
• this pre-compensation carried out from the encoding, ensures that the transmitted data are not divergent for the low frequencies.
• the distance p then represents a distance between a near virtual source M and the point O representing the origin of the spherical base of FIG. 3.
• a first near field simulation filter is thus applied to simulate the presence of a virtual source at the distance p described above.
• the terms of the coefficient of this filter diverge in the low frequencies ( Figure 6) and, on the other hand, the distance p above will not necessarily represent the distance between the high- speakers of a playback device and a point P of perception (figure 7).
• a pre-compensation is applied to the encoding, putting in
• the near-field pre-compensation of the loudspeakers (placed at the distance R), at the encoding stage, can be combined with a simulated near-field effect of a virtual source placed at a distance p.
• a total filter is finally brought into play resulting, on the one hand, from the near field simulation, and, on the other hand, from the near field compensation, the coefficients of this filter being able to be expressed analytically by the relation:
• the total filter given by the relation [Ail] is stable and constitutes the "distance encoding" part in the spatial ambisonic encoding according to the invention, as represented in FIG. 8.
• the coefficients of these filters correspond to functions of monotonic transfer of the frequency, which tend towards the value 1 in • high frequencies and towards the value (R / p) m in low frequencies.
• R between an auditory perception point and the HPi loudspeakers is effectively of the order of one or a few meters.
• total filters are also provided (near field compensation and, if necessary, simulation of a near field) H m ⁇ pc ' c ' (co) which are applied to the ambisonic components, according to their order m, to carry out the encoding of the distance, as represented in FIG. 8.
• H m ⁇ pc ' c ' (co) which are applied to the ambisonic components, according to their order m, to carry out the encoding of the distance, as represented in FIG. 8.
• FIG. 11B the propagation of the initial sound wave has been represented from a near-field source situated at a distance p from a point in the acquisition space which corresponds, in the reproduction space , at point P of Figure 7 of auditory perception.
• the listeners symbolized by diagrammed heads
• an advantageous method for defining a digital filter from the analytical expression of this filter in the analog domain in continuous time consists of a
• ⁇ p / c (c being the acoustic speed in the environment, typically 340 m / s in the air).
• the bilinear transform consists in presenting, for a sampling frequency f s , the relation [Ail] in the form:
• X m7q are the q successive roots of the Bessel polynomial
• q l and are expressed in table 1 below, for different orders m, in the respective forms of their real part, their modulus (separated by a comma) and their value (real) when m is odd.
• Table 1 values P e X m , q ⁇ ⁇ X m, q (and R e ⁇ . X m, m ⁇ when m is odd) of a Bessel polynomial calculated using the MATLAB ⁇ calculation software.
• Digital filters are thus produced in the form of an infinite impulse response, easily configurable as shown above. It should be noted that an implementation in the form of a finite impulse response can be envisaged and consists in calculating the complex spectrum of the transfer function from the analytical formula, then to deduce a finite impulse response by inverse Fourier transform. We then apply a convolution operation for filtering.
• FIG. 8 a modified ambisonic representation is defined (FIG. 8), by adopting as transmissible representation of the signals expressed in the frequency domain, in the form:
• R is a reference distance with which a compensated near field effect is associated and c is the speed of sound (typically 340 m / s in air).
• c is the speed of sound (typically 340 m / s in air).
• modified ambisonics has the same scalability properties (schematically represented by transmitted data "surrounded” near the arrow TR in figure 1) and obeys the same rotational transformations of the field (module 4 in figure 1) as the usual ambisonic representation .
• the decoding operation is adaptable to any restitution device, of radius R 2 , different from the reference distance R above.
• type filters are applied H ( ⁇ ) ⁇ as described above, but with distance parameters R and R 2 , instead of p and R.
• R / c is to be memorized (and / or transmit) between encoding and decoding.
• the filter module shown there is provided for example in a processing unit of a rendering device.
• the ambisonic components received were pre-compensated for encoding for a reference distance Ri as the second distance.
• the reproduction device comprises a plurality of loudspeakers arranged at a third distance R 2 from an auditory perception point P, this third distance R 2 being different from the aforementioned second distance R_.
• the invention also makes it possible to mix several ambisonic representations of sound fields (real and / or virtual sources), the reference distances R of which are different (if necessary with infinite reference distances and corresponding to distant sources).
• the reference distances R are different (if necessary with infinite reference distances and corresponding to distant sources).
• we will filter a pre-compensation of all these sources at the smallest reference distance, before mixing the signals ambisonics, which allows the reproduction to obtain a correct definition of the sound relief.
• the distance encoding with near field pre-compensation is advantageously applied in combination with the focusing processing.
• the wave emitted by each loudspeaker is defined by a prior processing of "re-encoding" of the ambisonic field at the center of the restitution device, as follows.
• decoding consists in comparing the original ambisonic signals received by the restitution device, in the form:
• the number of loudspeakers is greater than or equal to the number of ambisonic components to be decoded and the decoding matrix D is expressed, as a function of the re-encoding matrix C, in the form:
• the matrixing operation is preceded by a filtering operation which compensates for the near field on each component B mn ⁇ and which can be implemented in digital form, as described above, with reference to the relation [A14] .
• the "re-encoding" matrix C is specific to the restitution device. Its coefficients can be determined initially by configuration and sound characterization of the reproduction device reacting to a predetermined excitation.
• the decoding matrix D is also specific to the restitution device. Its coefficients can be determined by the relation [B8]. Using the previous notation where B is the matrix of precompensated ambisonic components, these can be transmitted to the restitution device in matrix form B with:
• the restitution device then decodes the data received in matrix form B (column vector of the transmitted components) by applying the decoding matrix D to the pre-compensated ambisonic components, to form the signals Si intended to supply the loudspeakers HPi, with:
• FIG. 13A a listener having a headset with two headphones of a binaural synthesis device is represented.
• the two ears of the listener are arranged at respective points 0 L (left ear) and 0 R (right ear) in space.
• the center of the listener's head is arranged at point O and the radius of the listener's head is a value.
• a sound source must be heard hearing at a point M in space, located at a distance r from the center of the listener's head (and respectively at distances r R from the right ear and r L from the ear left).
• the direction of the source placed at point M is defined by the vectors F, FR and f j _.
• binaural synthesis is defined as follows.
• Each listener has his own ear shape.
• the perception of a sound in space by this listener is done by learning, from birth, according to the shape of the ears (in particular the shape of the flags and the dimensions of the head) specific to this listener.
• the perception of a sound in space is manifested inter alia by the fact that the sound reaches one ear, before the other ear, which results in a delay ⁇ between the signals to be emitted by each earpiece of the listening device.
• restitution applying binaural synthesis.
• the restitution device is initially configured, for the same listener, by scanning a sound source around its head, at the same distance R from the center of its head. It will thus be understood that this distance R can be considered as a distance between a "restitution point" as stated above and a hearing perception point (here the center O of the listener's head).
• the index L is associated with the signal to be restored by the earpiece attached to the left ear and the index R is associated with the signal to be restored by the earpiece attached to the right ear.
• a delay is applied to the initial signal S for each channel intended to produce a signal for a separate listener.
• These delays ⁇ L and ⁇ R are a function of a maximum delay X MAX which here corresponds to the ratio a / c where a, as indicated previously, corresponds to the radius of the listener's head and c to the speed of sound.
• these delays are defined as a function of the difference in distance from point O (center of the head) to point M (position of the source whose sound is to be reproduced, in FIG. 13A) and of each ear at this point
• respective gains S and -R / are also applied to each channel, which are a function of a ratio of the distances from point O to point M and of each ear to point M.
• Respective modules applied to each channel 2 L and 2 R encode the signals of each channel, in an ambisonic representation, with NFC near field pre-compensation (for "jVear Field Compensation") within the meaning of present invention.
• the signals originating from the source M are transmitted to the restitution device comprising ambisonic decoding modules, for each channel, 5 L and 5 R .
• the restitution device comprising ambisonic decoding modules, for each channel, 5 L and 5 R .
• an ambisonic encoding / decoding is applied, with near field compensation, for each channel (left earpiece, right earpiece) in the reproduction with binaural synthesis (here of "B-FORMAT" type), in split form.
• the near field compensation is carried out, for each channel, with a distance r L and r R as the first distance between each ear and the position M of the sound source to be restored.
• a microphone 141 comprises a plurality of transducer capsules, capable of picking up acoustic pressures and reproducing electrical signals Si, ..., S N.
• CAPi capsules are arranged on a sphere of predetermined radius r (here, a rigid sphere, such as a ping-pong ball for example). The capsules are spaced evenly on the sphere. In practice, we choose the number N of capsules as a function of the order M desired for the ambisonic representation.
• the near-field pre-compensation can be applied not only for the simulation of virtual source, as indicated above, but also to the acquisition and, more generally, by combining the pre-compensation of field close to all types of treatment involving ambisonic representation.
• EQ ra is an equalizing filter which compensates for a weighting m which is linked to the directivity of the capsules and which also includes diffraction by the rigid sphere.
• this equalization filter is not stable and an infinite gain is obtained at very low frequencies.
• the spherical harmonic components themselves, are not of finite amplitude when the sound field is not limited to a propagation of plane waves, i.e. from from distant sources, as we saw earlier.
• the signals Si to S N are recovered from the microphone 141. If necessary, a pre-equalization of these signals is applied by a processing module 142.
• the module 143 makes it possible to express these signals in the ambisonic context, in the form matrix.
• Module 144 applies the filter of relation [C7] to the components ambisonics expressed as a function of the radius r of the sphere of the microphone 141.
• the near field compensation is performed for a reference distance R as the second distance.
• the signals encoded and thus filtered by the module 144 can be transmitted, if necessary, with the parameter representative of the reference distance R / c.
• the near field compensation within the meaning of the present invention can be applied to all types of processing involving an ambisonic representation.
• This near field compensation makes it possible to apply the ambisonic representation to a multiplicity of sound contexts where the direction of a source and advantageously its distance must be taken into account.
• the possibility of representing sound phenomena of all types (near or far fields) in the ambisonic context is ensured by this pre-compensation, due to the limitation to finite real values of the ambisonic components.
• the present invention is not limited to the embodiment described above by way of example; it extends to other variants.
• the near field pre-compensation can be integrated, in the encoding, as much for a near source as for a far source.
• the distance p expressed above will be considered to be infinite, without substantially modifying the expression of the filters H m given above.
• processing using room effect processors which generally provide decorrelated signals usable for modeling the late diffuse field (late reverberation) can be combined with near-field pre-compensation.
• the encoding principle within the meaning of the present invention can be generalized to radiation models other than monopolar sources (real or virtual) and / or loudspeakers.
• any form of radiation in particular a source spread out in space
• any form of radiation can be expressed by integration of a continuous distribution of point elementary sources.
• the present invention applies to all types of sound spatialization systems, in particular for "virtual reality” type applications (navigation in virtual scenes in three-dimensional space, games with three-dimensional sound spatialization, "chat” type conversations with sound on the Internet), interface sonifications, audio editing software for recording, mixing and playing music, but also for acquisition, from the use of three-dimensional microphones, for taking musical or cinematographic sound, or for transmitting sound environment on the Internet, for example for sound-activated "WebCam”.
• “virtual reality” type applications novigation in virtual scenes in three-dimensional space, games with three-dimensional sound spatialization, "chat” type conversations with sound on the Internet
• interface sonifications audio editing software for recording, mixing and playing music, but also for acquisition, from the use of three-dimensional microphones, for taking musical or cinematographic sound, or for transmitting sound environment on the Internet, for example for sound-activated "WebCam”.

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