EP1563485B1 - Verfahren zur verarbeitung von audiodateien und erfassungsvorrichtung zur anwendung davon - Google Patents

Verfahren zur verarbeitung von audiodateien und erfassungsvorrichtung zur anwendung davon Download PDF

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EP1563485B1
EP1563485B1 EP03782553A EP03782553A EP1563485B1 EP 1563485 B1 EP1563485 B1 EP 1563485B1 EP 03782553 A EP03782553 A EP 03782553A EP 03782553 A EP03782553 A EP 03782553A EP 1563485 B1 EP1563485 B1 EP 1563485B1
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distance
sound
components
point
signals
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EP1563485A1 (de
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Jérôme DANIEL
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Orange SA
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France Telecom SA
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    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10HELECTROPHONIC 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
    • G10H1/00Details of electrophonic musical instruments
    • G10H1/0091Means for obtaining special acoustic effects
    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10HELECTROPHONIC 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
    • G10H1/00Details of electrophonic musical instruments
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04SSTEREOPHONIC SYSTEMS 
    • H04S2400/00Details of stereophonic systems covered by H04S but not provided for in its groups
    • H04S2400/15Aspects of sound capture and related signal processing for recording or reproduction
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04SSTEREOPHONIC SYSTEMS 
    • H04S2420/00Techniques used stereophonic systems covered by H04S but not provided for in its groups
    • H04S2420/11Application 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 the three-dimensional space implement methods for processing the audio signal applied to the simulation of acoustic and psychoacoustic phenomena. .
  • Such processing methods provide spatial encoding of the acoustic field, its transmission and spatial reproduction on a set of loudspeakers or on headphones of a stereo headset.
  • a first category of treatments relates to processes for synthesizing the room effect, or more generally for environmental effects. From a description of one or more sound sources (transmitted signal, position, orientation, directivity, or other) and based on a room effect model (involving a room geometry, or an acoustic perception desired), a set of elementary acoustic phenomena (direct waves, reflected or diffracted), or a macroscopic acoustic phenomenon (reverberated and diffuse field), allowing to translate the spatial effect at the level of a listener located at a selected point of auditory perception, in the three-dimensional space.
  • sound sources transmitted signal, position, orientation, directivity, or other
  • a room effect model involving a room geometry, or an acoustic perception desired
  • a set of elementary acoustic phenomena direct waves, reflected or diffracted
  • a macroscopic acoustic phenomenon reverberated and diffuse field
  • a set of signals typically associated with reflections ( "secondary" sources , active by re-emission of a received main wave, having a spatial position attribute) and / or associated with a late reverberation (decorrelated signals for a field) are then calculated. diffuse).
  • a second category of processes relates to the positional or directional rendering of sound sources. These methods are applied to signals determined by a method of the first category described above (involving primary and secondary sources) depending on the spatial description (source position) associated with them.
  • these methods according to this second category make it possible to obtain signals to be broadcast on loudspeakers or headphones, in order to finally give to a listener the auditory impression of sound sources placed at predetermined respective positions around the auditor.
  • Processes according to this second category are called "creators of three-dimensional sound images", because of the distribution in the three-dimensional space of the feeling of the position of the sources by a listener.
  • Methods according to the second category generally comprise a first stage of spatial encoding of the elementary acoustic events which produces a representation of the sound field in the three-dimensional space.
  • this representation is transmitted or stored for a deferred use.
  • the decoded signals are delivered on loudspeakers or headphones of a playback device.
  • the present invention is rather in 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 spatialized conference, or others), an "acoustic" encoding of a natural sound field, during sound recording by one or more three-dimensional microphone networks.
  • virtual sound sources
  • acoustic encoding of a natural sound field
  • a similar acoustic encoding method is presented byJ. Chen et al: "Synthesis of 3D virtual auditory space via a spatial feature extraction and regularization model", Proceedings of the virtual reality annual international symposium, Seattle, Sept. 18-22, 1993, IEEE, New York, US, pages 188 -193.
  • the ambisonic encoding which will be described in detail below, consists of representing signals relating to one or more sound waves in a base of spherical harmonics (in spherical coordinates involving in particular an elevation angle and an azimuthal angle, characterizing a direction of the sound or sounds).
  • the components representing these signals and expressed in this basis of spherical harmonics are also a function, for the 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 particularly, this dependence of distance is expressed as a function of the sound frequency, as will be seen below.
  • this document presents a horizontal network of sensors, which assumes that the acoustic phenomena considered 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 acoustic 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 for encoding 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 sound recording of natural acoustic fields using three-dimensional microphone networks.
  • the data encoded and filtered in steps a) and b) are transmitted to the rendering device with a parameter representative of said second distance.
  • the rendering device comprising means for reading a memory medium
  • the encoded and filtered data is stored on a memory medium intended to be read by the rendering 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 whose coefficients are a function of said second and third distances.
  • the coefficients of a digital audio filter are defined from the root digital values of said power polynomials m.
  • the aforementioned polynomials are Bessel polynomials.
  • a microphone comprising an acoustic transducer array arranged substantially on the surface of a sphere whose center corresponds substantially to said reference point, to obtain said signals representative of at least one sound. propagating in three-dimensional space.
  • a global filter is applied to step b) in order firstly to compensate for a near-field effect as a function of said second distance and secondly to equalize the signals from the transducers to compensate for a difference. directivity weighting of said transducers.
  • a number of transducers function of a selected total number of components to represent the sound in said base of spherical harmonics.
  • a total number of components in the base of spherical harmonics to obtain, at the restitution, a region of the space around the point of perception in which the sound reproduction is faithful and whose dimensions are increasing with the total number of components.
  • a playback device having a number of speakers at least equal to said total number of components.
  • the filtering performed by the processing unit consists, on the one hand, of equalizing, as a function of the radius of the sphere, the signals coming from the transducers to compensate for a directivity weighting of said transducers and, on the other hand, to compensating a near-field effect according to said reference distance.
  • FIG. 1 represents by way of illustration a global system of sound spatialization.
  • a simulation module of a virtual scene defines a sound object as a virtual source of a signal, for example monophonic, of position chosen in the three-dimensional space and which defines a direction of sound.
  • specifications of the geometry of a virtual room can be provided to simulate reverberation of the sound.
  • a processing module 11 applies a 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 for simulating reverberations or the like by applying delays and / or routine filtering.
  • the signals thus constructed are transmitted to a spatial encoding module 2a of the elementary contributions of the sources.
  • a natural sound recording can be performed as part of a sound recording by one or more microphones arranged in a chosen manner with respect to 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 to the signals generated by the module 1a and encoded by the module 2a (from the virtual sources).
  • the mixed signals are then transmitted or stored on a medium, for later retrieval (arrow TR). They are then applied to a decoding module 5, for the purpose of rendering on a reproduction device 6 comprising loudspeakers.
  • the decoding step 5 may be preceded by a step of manipulation of the sound field, for example by rotation, by means of a processing module 4 provided upstream of the decoding module 5.
  • the reproduction device may be in the form of a multiplicity of loudspeakers, arranged for example on the surface of a sphere in a three-dimensional configuration (periphery) to ensure, in the restitution, in particular a sense of direction sound in three-dimensional space.
  • an auditor place generally in the center of the sphere formed by the network of speakers, this center corresponding to the auditory perception point cited above.
  • the speakers of the playback device can be arranged in a plane (two-dimensional panoramic configuration), the speakers being arranged in particular on a circle and the listener usually placed in the center of this circle.
  • the rendering device may be in the form of a "surround" type device (5.1).
  • the rendering device can be in the form of a headset with two earphones for a binaural synthesis of the sound reproduced, which allows the listener to feel a direction of the sources in the three-dimensional space, as will be discussed in more detail below.
  • a two-speaker reproduction device for a feeling in the three-dimensional space, can also be in the form of a transaural restitution device, with two loudspeakers arranged at a selected distance from a listener.
  • a 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 as well defined in terms of incidence (direction of the source as seen by the listener) and 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 a sound reproduction in the three-dimensional space, as indicated above with reference to FIG.
  • the set of weighting factors B mn ⁇ 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 sound (or pressure field) in the spherical harmonics basis Y mn ⁇ .
  • the spherical harmonics form an orthonormal basis where the scalar products between harmonic components and, generally between two functions F and G, are respectively defined by: ⁇ Y m not ⁇
  • Y m ' not ' ⁇ ' > 4 ⁇ ⁇ mm ' ⁇ nn ' ⁇ ⁇ ' .
  • BOY WUT 4 ⁇ 1 4 ⁇ ⁇ F ( ⁇ , ⁇ ) BOY WUT ( ⁇ , ⁇ ) , d ⁇ ( ⁇ , ⁇ )
  • Spherical harmonics are bounded real 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 spherical harmonic functions.
  • the radial functions j m (kr) are spherical Bessel functions, the module of which is illustrated for some values of the order m in FIG.
  • ambisonic representation can be given by a base of spherical harmonics as follows.
  • the ambisonic components of the same order m finally express “derivatives” or “moments” of order m of the pressure field in the vicinity of the origin O (center of the sphere shown in FIG. 3).
  • B 11 + 1 X
  • B 11 - 1 Y
  • an ambisonic system takes into account a subset of spherical harmonic components, as described above.
  • a system of order M when this one takes into account ambisonic components of subscript m ⁇ M.
  • the rendering device comprises loudspeakers disposed on the surface of a sphere (" periphery "), it is possible in principle to use as many harmonics as there are loudspeakers.
  • the reference S designates the pressure signal carried by a plane wave and picked up at the point O corresponding to the center of the sphere of FIG. 3 (origin of the base in spherical coordinates).
  • the incidence of the wave is described by the azimuth ⁇ and the elevation ⁇ .
  • a filter is applied F m ( ⁇ / vs ) to "bend" the shape of the wave fronts, considering that a near field emits, as a first approximation, a spherical wave.
  • this additional filter is of the "integrator" type, with an amplifying effect increasing and diverging (unbounded) as the sound frequencies decrease towards zero.
  • a pre-compensation of the near field is introduced at the very stage of the encoding, this compensation involving filters of the analytical form.
  • amplification F m ( ⁇ / vs ) ( ⁇ ) whose effect appears in FIG. 6 is compensated by attenuation of the filter applied as soon as encoding 1 F m ( R / vs ) ( ⁇ ) .
  • the coefficients of this compensation filter 1 F m ( R / vs ) ( ⁇ ) are increasing with the frequency of the sound and, in particular, tend towards zero, for the low frequencies.
  • this pre-compensation, performed as soon as encoding ensures that the data transmitted are not divergent for low frequencies.
  • a pre-compensation is applied to the encoding, involving a filter of the type 1 F m ( R / vs ) ( ⁇ ) as indicated above, which allows, on the one hand, to transmit bounded signals, and, on the other hand, to choose the distance R, from the encoding, for the restitution of the sound from the loudspeakers HP i , as shown in FIG. 7.
  • a virtual source placed at the distance p of the origin O was simulated at the time of acquisition (FIG.
  • the pre-compensation of the near field of the loudspeakers (placed at the distance R), at the stage of the encoding, can be combined with a simulated near-field effect of a virtual source placed at a distance p.
  • a total filter ultimately comes into play resulting, on the one hand, from the simulation of the near field, and, on the other hand, from the compensation of the near field, the coefficients of this filter being able to express itself.
  • H m NFC ( ⁇ / vs , R / vs ) ( ⁇ ) F m ( ⁇ / vs ) ( ⁇ ) F m ( R / vs ) ( ⁇ )
  • the total filter given by the relation [A11] 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 the functions of FIG. monotonic transfer of the frequency, which tend towards the value 1 in high frequencies and towards the value (R / ⁇ ) m in low frequencies.
  • the distance R between an auditory perception point and the speakers HP i is actually of the order of one or a few meters.
  • steps a) and b) above can be brought together in one and the same global step, or even be interchanged (with distance encoding and compensation filtering, followed by direction encoding).
  • the method according to the invention is therefore not limited to a successive implementation over time of steps a) and b).
  • FIG. 11B shows the propagation of the initial sound wave from a near-field source situated at a distance p from a point in the acquisition space that corresponds, in the restitution space at point P of Figure 7 of auditory perception. Note in FIG. 11A that the listeners (symbolized by schematized heads) can locate the virtual source in the same geographical location located at the distance p from the perception point P in FIG. 11B.
  • H m NFC ( ⁇ / vs , R / vs ) ( ⁇ ) F m ( ⁇ / vs ) ( ⁇ ) F m ( R / vs ) ( ⁇ )
  • Table 1 values ⁇ i> R ⁇ / i> ⁇ sub> ⁇ i> e ⁇ / i> ⁇ /sub>[ ⁇ i> X ⁇ / i> ⁇ sub> ⁇ i> m, q ⁇ / i> ⁇ / sub>],
  • the digital filters are thus implemented from the values of Table 1, by providing cascades of cells of order 2 (for m even), and an additional cell (for odd m), from the relationships [A14] given here. -before.
  • Digital filters are thus produced in an infinite impulse response form, which is easily parameterizable as shown above. It should be noted that an implementation in a finite impulse response form can be envisaged and consists in calculating the complex spectrum of the transfer function from the analytic formula, then deduce a finite impulse response by inverse Fourier transform. A convolution operation is then applied for filtering.
  • 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).
  • This modified ambisonic representation has the same scalability properties (schematically represented by transmitted data "surrounded" near the arrow TR of FIG. 1) and obeys the same transformations of rotation of the field (module 4 of FIG. usual ambisonic.
  • the decoding operation is adaptable to any rendering device, of radius R 2 , different from the reference distance R above.
  • filters of the type H m NFC ( ⁇ / vs , R / vs ) ( ⁇ ) as described above, but with distance parameters R and R 2 , instead of p and R.
  • R / c is to be memorized (and / or transmitted) between encoding and decoding.
  • the filtering module represented therein is provided, for example, in a processing unit of a rendering device.
  • Ambisonic components received were pre-compensated for encoding for a reference distance R 1 as a second distance.
  • the rendering 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 second aforementioned distance R 1 .
  • the filtering module of FIG. 12, in the form H m NFC ( R 1 / vs , R two / vs ) ( ⁇ ) then adapts, upon reception of the data, the pre-compensation at the distance R 1 for a reproduction at the distance R 2 .
  • the rendering device also receives the parameter R 1 / c.
  • the invention also makes it possible to mix several ambisonic representations of sound fields (real and / or virtual sources), whose reference distances R are different (where appropriate with infinite reference distances and corresponding to distant sources).
  • a pre-compensation of all these sources will be filtered at a smallest reference distance, before mixing the signals ambisic, which allows the restitution to obtain a correct definition of the sound relief.
  • the distance encoding with near-field pre-compensation is advantageously applied in combination with the focus processing.
  • the wave transmitted by each speaker is defined by a processing prior to "re-encoding" the ambisonic field in the center of the rendering device, as follows.
  • the wave emitted by a loudspeaker of index i and incidence ( ⁇ i and ⁇ i ) is powered by a signal Si.
  • This loudspeaker participates in the reconstruction of the component B mn ' , by his contribution S i .
  • Y mn ⁇ ( ⁇ i , ⁇ i ) the wave emitted by a loudspeaker of index i and incidence ( ⁇ i and ⁇ i ) is powered by a signal Si.
  • vs i [ Y 00 + 1 ( ⁇ i , ⁇ i ) Y 11 + 1 ( ⁇ i , ⁇ i ) Y 11 - 1 ( ⁇ i , ⁇ i ) ⁇ Y m not ⁇ ( ⁇ i , ⁇ i ) ⁇ ]
  • the relation [B4] thus defines a re-encoding operation, prior to the restitution.
  • decoding verifying different criteria by frequency bands is possible, which makes it possible to offer an optimized reproduction according to the listening conditions, in particular with regard to the positioning constraint at the center. O of the sphere of Figure 3, during the restitution.
  • the mastering operation is preceded by a filtering operation that compensates for the near field on each component.
  • B mn ⁇ and which can be implemented in digital form, as described above, with reference to relation [A14].
  • the matrix C of "re-encoding" is specific to the rendering device. Its coefficients can be determined initially by parameterization and sound characterization of the restitution device reacting to a predetermined excitation.
  • a listener having a two-headset headset of a binaural synthesis device is shown.
  • the two ears of the listener are arranged at respective points O L (left ear) and O R (right ear) of the space.
  • the center of the listener's head is located at point O and the radius of the listener's head is of value a.
  • a sound source must be audibly perceived at a point M in the space, 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 at the point M is defined by the vectors r ⁇ , r ⁇ R and r ⁇ The .
  • binaural synthesis is defined as follows.
  • Each listener has an ear shape of its own.
  • the perception of a sound in the space by this listener is done by learning, from birth, according to the form of the ears (in particular the shape of the pavilions and the dimensions of the head) peculiar to this listener.
  • the perception of 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 earphone of the device. restitution applying binaural synthesis.
  • the playback device is initially set, for the same listener, by scanning a sound source around his head, at the same distance R from the center of his head. It will be understood that this distance R can be considered as a distance between a "restitution point" as stated above and a point of auditory perception (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 for each channel for producing a signal for a separate earphone is applied to the initial signal S.
  • These delays ⁇ L and ⁇ R are a function of a maximum delay ⁇ MAX which corresponds here 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 the point O (center of the head) to the point M (position of the source whose sound is to be restored, in FIG. 13A) and of each ear at this point. M.
  • respective gains g L and g R are also applied to each channel, which are a function of a ratio of the distances from the point O to the point M and from each ear to the point M.
  • the respective modules applied to each channel 2 L and 2 R encode the signals of each channel, in an ambisonic representation, with near field pre-compensation NFC (for "Near Field Compensation") in the sense of the present invention.
  • the signals coming from the source M are transmitted to the reproduction device comprising ambisonic decoding modules, for each channel, 5 L and 5 R.
  • the reproduction device comprising ambisonic decoding modules, for each channel, 5 L and 5 R.
  • an ambisonic encoding / decoding, with near-field compensation is applied for each channel (left listener, right listener) in the binaural synthesis restitution (here of type "B-FORMAT"), in split form.
  • the near-field compensation is effected, for each channel, with the first distance p a distance r L and r R 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 S l , ..., S N.
  • Caps CAP i 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 with a regular pitch on the sphere. In practice, the number N of capsules is chosen according to the desired order M for the ambisonic representation.
  • the pre-compensation of the near field can be applied not only for the virtual source simulation, as indicated above, but also to the acquisition and, more generally, by combining the pre-compensation of field close to all types of treatments involving an ambisonic representation.
  • EQ m is an equalizer filter that compensates a weighting W m which is related to the directivity of the capsules and which further includes the diffraction by the rigid sphere.
  • this equalization filter is not stable and we obtain an infinite gain at very low frequencies.
  • the spherical harmonic components themselves, are not of finite amplitude when the sound field is not limited to propagation of plane waves, that is to say from distant sources, as we saw earlier.
  • the signals S 1 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, under matrix form.
  • the module 144 applies the filter of the relation [C7] to the components ambisonic 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 a 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.
  • 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 the representation of sound phenomena of all types (near or far fields) in the ambisonic context is ensured by this pre-compensation, because of the limitation to finite real values of the ambison components.
  • the near-field pre-compensation can be integrated, at the encoding, as much for a near source as for a distant source.
  • the distance p expressed above will be considered infinite, without substantially modifying the expression of the filters H m given above.
  • processing using room effect processors that typically provide decorrelated signals that can be used to model the late diffuse field (late reverberation) can be combined with near field pre-compensation.
  • the various spherical harmonic components (with a chosen order M) can then be constructed by applying a gain correction for each ambisonic component and a field compensation close to the loudspeakers (with a reference distance R separating the loudspeakers the point of auditory perception as shown in Figure 7).
  • the encoding principle in the sense of the present invention is generalizable to radiation models other than monopolar sources (real or virtual) and / or speakers.
  • any form of radiation can be expressed by integration of a continuous distribution of point elementary sources.
  • a decoding method has been described above in which a matrix system involving the ambison components is applied.
  • it may be provided a generalized processing by fast Fourier transforms (circular or spherical) to limit the computing time and computing resources (in terms of memory) necessary for the decoding process.
  • the pre-compensation encoding method may be coupled to a digital audio compression for quantizing and adjusting the gain for each frequency subband.
  • the present invention applies to all types of sound spatialization systems, especially for "virtual reality” type applications (navigation in virtual scenes in three-dimensional space, cat-type conversations on the Internet), interface sonification, audio editing software for recording, mixing and restoring music, but also for acquiring, from use of three-dimensional microphones, for taking musical or cinematic sound, or for the transmission of sound environment on the Internet, for example for "Webcam” sound.

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  • Engineering & Computer Science (AREA)
  • Acoustics & Sound (AREA)
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  • Measurement Of Mechanical Vibrations Or Ultrasonic Waves (AREA)
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Claims (22)

  1. Verfahren zur Verarbeitung von Audiodateien, bei dem:
    a) man Signale kodiert, die mindestens einen Ton darstellen, der sich im dreidimensionalen Raum ausbreitet und von einer Quelle kommt, die in einem ersten Abstand (p) von einem Bezugspunkt (O) gelegen ist, um eine Darstellung des Tons durch in einer Basis von sphärischen Harmonischen ausgedrückte Komponenten (Bmn σ) mit einem diesem Bezugspunkt (O) entsprechenden Ursprung zu erhalten,
    b) und man an diese Komponenten (Bmn σ) eine Kompensierung eines Nahfeldeffekts durch eine Filterung anlegt, die eine Funktion von einem zweiten Abstand (R) ist, der bei einer Wiedergabe des Tons durch eine Wiedergabevorrichtung im Wesentlichen einen Abstand zwischen einem Wiedergabepunkt (Hpi) und einem Hörwahrnehmungspunkt (P) definiert.
  2. Verfahren nach Anspruch 1, bei dem, wenn diese Quelle vom Bezugspunkt (O) entfernt ist,
    - man Komponenten von aufeinanderfolgenden Ordnungen m bei der Darstellung des Tons in dieser Basis von sphärischen Harmonischen erhält und
    - man ein Filter (1/Fm) anlegt, dessen jeweils an eine Komponente der Ordnung m angelegte Koeffizienten sich analytisch in der Form des Kehrwerts eines Polynoms der Potenz m ausdrückt, dessen Variable umgekehrt proportional zur Tonfrequenz und zum zweiten Abstand (R) ist, um einen Nahfeldeffekt auf Höhe der Wiedergabevorrichtung zu kompensieren.
  3. Verfahren nach Anspruch 1, bei dem, wenn diese Quelle eine in diesem ersten Abstand (ρ) vorgesehene virtuelle Quelle ist,
    - man Komponenten von aufeinanderfolgenden Ordnungen m bei der Darstellung des Tons in dieser Basis von sphärischen Harmonischen erhält und
    - man ein Globalfilter (Hm) anlegt, dessen jeweils an eine Komponente der Ordnung m angelegte Koeffizienten sich analytisch in der Form eines Bruchs ausdrücken, dessen
    - Zähler ein Polynom der Potenz m ist, dessen Variable umgekehrt proportional zur Tonfrequenz und zum ersten Abstand (p) ist, um einen Nahfeldeffekt der virtuellen Quelle zu simulieren, und
    - dessen Nenner ein Polynom der Potenz m ist, dessen Variable umgekehrt proportional zu der Tonfrequenz und zu dem zweiten Abstand (R) ist, um den Nahfeldeffekt der virtuellen Quelle in den niedrigen Tonfrequenzen zu kompensieren.
  4. Verfahren nach einem der vorhergehenden Ansprüche, bei dem man die in den Schritten a) und b) kodierten und gefilterten Daten mit einem diesen zweiten Abstand darstellenden Parameter (R/c) zur Wiedergabevorrichtung überträgt.
  5. Verfahren nach einem der Ansprüche 1 bis 3, bei dem man, wenn die Wiedergabevorrichtung Mittel zum Lesen eines Speicherträgers umfasst, auf einem Speicherträger, der dazu bestimmt ist, von der Wiedergabevorrichtung gelesen zu werden, die in den Schritten a) und b) kodierten und gefilterten Daten mit einem diesen zweiten Abstand darstellenden Parameter (R/c) speichert.
  6. Verfahren nach einem der Ansprüche 4 und 5, bei dem man vor einer Tonwiedergabe durch eine Wiedergabevorrichtung, die eine Vielzahl von Lautsprechern umfasst, die in einem dritten Abstand (R2) von diesem Hörwahrnehmungspunkt (P) angeordnet sind, an die kodierten und gefilterten Daten ein Anpassungsfilter (Hm (R1/c,R2/c)) anlegt, dessen Koeffizienten eine Funktion von dem zweiten Abstand (R1) und dem dritten Abstand (R2) sind.
  7. Verfahren nach Anspruch 6, bei dem die jeweils an eine Komponente der Ordnung m angelegten Koeffizienten des Anpassungsfilters (Hm (R1/c,R2/c)) sich analytisch in der Form eines Bruchs ausdrücken, dessen
    - Zähler ein Polynom der Potenz m ist, dessen Variable umgekehrt proportional zur Tonfrequenz und zum zweiten Abstand (R) ist, und
    - dessen Nenner ein Polynom der Potenz m ist, dessen Variable umgekehrt proportional zu der Tonfrequenz und zu dem dritten Abstand (R2) ist.
  8. Verfahren nach einem der Ansprüche 2, 3 und 7, bei dem man für die Durchführung des Schritts b)
    - für Komponenten geradzahliger Ordnung m audionumerische Filter in der Form einer Kaskade von Zellen der Ordnung zwei und
    - für Komponenten ungeradzahliger Ordnung m audionumerische Filter in der Form einer Kaskade von Zellen der Ordnung zwei und eine zusätzliche Zelle der Ordnung eins vorsieht.
  9. Verfahren nach Anspruch 8, bei dem die Koeffizienten eines audionumerischen Filters bei einer Komponente der Ordnung m ausgehend von den numerischen Werten der Wurzeln dieser Polynome der Potenz m definiert sind.
  10. Verfahren nach einem der Ansprüche 2, 3, 7, 8 und 9, bei dem diese Polynome Bessel-Polynome sind.
  11. Verfahren nach einem der Ansprüche 1, 2 und 4 bis 10, bei dem man ein Mikrophon vorsieht, das ein Netz von akustischen Wandlern aufweist, die im Wesentlichen auf der Oberfläche einer Kugel angeordnet sind, deren Mittelpunkt im Wesentlichen dem Bezugspunkt (O) entspricht, um die Signale zu erhalten, die mindestens einen sich im dreidimensionalen Raum ausbreitenden Ton darstellen.
  12. Verfahren nach Anspruch 11, bei dem man im Schritt b) ein Globalfilter anlegt, um einerseits einen Nahfeldeffekt in Abhängigkeit von diesem zweiten Abstand (R) zu kompensieren und andererseits die von den Wandlern kommenden Signale zu egalisieren, um eine Richtwirkungsgewichtung der Wandler zu kompensieren.
  13. Verfahren nach einem der Ansprüche 11 und 12, bei dem man eine Anzahl von Wandlern vorsieht, die von einer gewählten Gesamtzahl von Komponenten abhängt, um den Ton in der Basis der sphärischen Harmonischen darzustellen.
  14. Verfahren nach einem der vorhergehenden Ansprüche, bei dem man im Schritt a) eine Gesamtzahl von Komponenten in der Basis der sphärischen Harmonischen wählt, um bei der Wiedergabe einen Bereich des Raums um den Wahrnehmungspunkt (P) herum zu erhalten, in dem die Wiedergabe des Tons getreu ist und dessen Abmessungen mit der Gesamtzahl von Komponenten zunehmen.
  15. Verfahren nach Anspruch 14, bei dem man eine Wiedergabevorrichtung vorsieht, die eine Anzahl von Lautsprechern von mindestens gleich der Gesamtzahl der Komponenten vorsieht.
  16. Verfahren nach einem der Ansprüche 1 bis 5 und 8 bis 13, bei dem
    - man eine Wiedergabevorrichtung vorsieht, die mindestens einen ersten und einen zweiten Lautsprecher vorsieht, die in einem gewählten Abstand von einem Hörer angeordnet sind,
    - für diesen Hörer eine Information der Empfindung der Lage von Tonquellen, die in einem vorbestimmten Bezugsabstand (R) vom Hörer gelegen sind, im Raum erhält und
    - die Kompensierung des Schritts b) mit dem Bezugsabstand im Wesentlichen als zweiten Abstand anlegt.
  17. Verfahren nach einem der Ansprüche 1 bis 3 und 8 bis 13 in Kombination mit einem der Ansprüche 4 und 5, bei dem
    - man eine Wiedergabevorrichtung vorsieht, die mindestens einen ersten und einen zweiten Lautsprecher umfasst, die in einem gewählten Abstand von einem Hörer angeordnet sind,
    - man für diesen Hörer eine Information der Empfindung der Lage von Tonquellen, die in einem vorbestimmten Bezugsabstand (R2) vom Hörer gelegen sind, im Raum erhält und
    - man vor einer Tonwiedergabe durch die Wiedergabevorrichtung an die in den Schritten a) und b) kodierten und gefilterten Daten ein Anpassungsfilter (Hm (R/c,R2/c) anlegt, dessen Koeffizienten eine Funktion des zweiten Abstands (R) und im Wesentlichen des Bezugsabstands (R2) sind.
  18. Verfahren nach einem der Ansprüche 16 und 17, bei dem
    - die Wiedergabevorrichtung einen Kopfhörer mit zwei Hörern für die Ohren des Hörers umfasst und
    - man getrennt für jeden Hörer die Kodierung und Filterung der Schritte a) und b) für Signale, die jeweils zur Versorgung jedes Hörers bestimmt sind, anlegt, mit als erstem Abstand (p) jeweils einen Abstand (rR, rL) der jedes Ohr von einem Standort (M) einer wiederzugebenden Quelle trennt.
  19. Verfahren nach einem der vorhergehenden Ansprüche, bei dem man in den Schritten a) und b) ein Matrixsystem in Form bringt, das mindestens umfasst:
    - eine Matrix (B), die die Komponenten in der Basis der sphärischen Harmonischen umfasst, und
    - eine diagonale Matrix (Diag(1/Fm)), deren Koeffizienten Filterkoeffizienten des Schritts b) entsprechen, und man die Matrizes multiplizierten, um eine resultierende Matrix von kompensierten Komponenten () zu erhalten.
  20. Verfahren nach Anspruch 19, bei dem
    - die Wiedergabevorrichtung eine Vielzahl von Lautsprechern umfasst, die in einem gleichen Abstand (R) vom dem Hörwahrnehmungspunkt (P) angeordnet sind, und,
    - um diese in den Schritten a) und b) kodierten und gefilterten Daten zu dekodieren und Signale zu formen, die dafür ausgelegt sind, die Lautsprecher zu speisen,
    * man ein Matrixsystem bildet, das die resultierende Matrix () und eine vorbestimmte, der Wiedergabevorrichtung zugeordnete Dekodierungsmatrix (D) umfasst, und
    * man eine Matrix (S), die die Signale zur Versorgung der Lautsprecher darstellende Koeffizienten aufweist, durch Multiplikation der Matrix () der kompensierten Komponenten mit der Dekodierungsmatrix (D) erhält.
  21. Tonerfassungsvorrichtung, umfassend ein Mikrophon, das mit einem Netz von akustischen Wandlern versehen ist, die im Wesentlichen auf der Oberfläche einer Kugel angeordnet sind, dadurch gekennzeichnet, dass sie außerdem eine Verarbeitungseinheit aufweist, die dafür ausgelegt ist,
    - jeweils von einem Wandler ausgehende Signale zu empfangen,
    - an diese Signale eine Kodierung anzulegen, um eine Darstellung des Tons durch Komponenten (Bmn σ), die in einer Basis von sphärischen Harmonischen ausgedrückt sind, mit einem dem Mittelpunkt dieser Kugel (O) entsprechenden Ursprung zu erhalten,
    - und an diese Komponenten (Bmn σ) eine Filterung anzulegen, die eine Funktion einerseits von einem dem Radius der Kugel (r) entsprechenden Abstand und andererseits von einem Bezugsabstand (R) ist.
  22. Vorrichtung nach Anspruch 21, dadurch gekennzeichnet, dass die Filterung darin besteht, dass einerseits in Abhängigkeit von dem Radius der Kugel die von den Wandlern kommenden Signale egalisiert werden, um eine Richtwirkungsgewichtung der Wandler zu kompensieren, und andererseits ein Nahfeldeffekt in Abhängigkeit von einem gewählten Bezugsabstand (R) kompensiert wird, der bei einer Wiedergabe des Tons im Wesentlichen einen Abstand zwischen einem Wiedergabepunkt (HPi) und einem Hörwahrnehmungspunkt (P) definiert.
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FR2847376B1 (fr) 2005-02-04
JP2006506918A (ja) 2006-02-23
FR2847376A1 (fr) 2004-05-21
US20060045275A1 (en) 2006-03-02
ES2261994T3 (es) 2006-11-16
BR0316718A (pt) 2005-10-18
ZA200503969B (en) 2006-09-27

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