EP2754151B2 - Dispositif, procédé et système électroacoustique de prolongement d'un temps de réverbération - Google Patents
Dispositif, procédé et système électroacoustique de prolongement d'un temps de réverbération Download PDFInfo
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- EP2754151B2 EP2754151B2 EP12756411.0A EP12756411A EP2754151B2 EP 2754151 B2 EP2754151 B2 EP 2754151B2 EP 12756411 A EP12756411 A EP 12756411A EP 2754151 B2 EP2754151 B2 EP 2754151B2
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- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10K—SOUND-PRODUCING DEVICES; METHODS OR DEVICES FOR PROTECTING AGAINST, OR FOR DAMPING, NOISE OR OTHER ACOUSTIC WAVES IN GENERAL; ACOUSTICS NOT OTHERWISE PROVIDED FOR
- G10K15/00—Acoustics not otherwise provided for
- G10K15/08—Arrangements for producing a reverberation or echo sound
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- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10K—SOUND-PRODUCING DEVICES; METHODS OR DEVICES FOR PROTECTING AGAINST, OR FOR DAMPING, NOISE OR OTHER ACOUSTIC WAVES IN GENERAL; ACOUSTICS NOT OTHERWISE PROVIDED FOR
- G10K15/00—Acoustics not otherwise provided for
- G10K15/08—Arrangements for producing a reverberation or echo sound
- G10K15/12—Arrangements for producing a reverberation or echo sound using electronic time-delay networks
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04S—STEREOPHONIC SYSTEMS
- H04S7/00—Indicating arrangements; Control arrangements, e.g. balance control
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- H—ELECTRICITY
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- H04S—STEREOPHONIC SYSTEMS
- H04S2420/00—Techniques used stereophonic systems covered by H04S but not provided for in its groups
- H04S2420/13—Application of wave-field synthesis in stereophonic audio systems
Definitions
- the present invention relates to a device, a method and an electroacoustic system for reverberation time extension.
- a room is not optimal for different applications from an acoustic point of view. So a musical performance usually requires some reverb to sound good. On the other hand, speakers are sometimes incomprehensible when the room is too reverberant. An adjustment of the reverberation time with the help of a public address system is therefore useful.
- electroacoustic systems for reverberation time extension can be used. Such systems can either be e.g. be retrofitted into an existing concert hall. Likewise, however, it may be useful to begin with the construction and construction of corresponding buildings and halls, e.g. in trade fair construction, to provide an electroacoustic system for reverberation time extension and to include it in the planning of the building. Also in the context of audio playback for entertainment purposes, a reverberation time extension may be desirable.
- Wave Field Synthesis was researched at the TU Delft and first presented in the late 1980s ( Berkhout, AJ; de Vries, D .; Vogel, P .: Acoustic control by wave-field synthesis. JASA 93, 1993 ).
- WFS Huygens' principle of wave theory: Every point, which is detected by a wave, is the starting point of an elementary wave, which spreads in a spherical or circular manner. Applied to the acoustics can be simulated by a large number of speakers, which are arranged side by side (a so-called speaker array), any shape of an incoming wavefront.
- a single point source to be reproduced and a linear arrangement of the speakers the audio signals of each speaker must be fed with a time delay and amplitude scaling so that the radiated sound fields of each speaker properly overlap. With multiple sound sources, the contribution to each speaker is calculated separately for each source and the resulting signals added together. In a room with reflective walls, reflections can also be reproduced as additional sources via the loudspeaker array. The cost of the calculation therefore depends heavily on the number of sound sources, the reflection characteristics of the recording room and the number of speakers.
- the advantage of this technique is in particular that a natural spatial sound impression over a large area of the playback room is possible.
- the direction and distance of sound sources are reproduced very accurately.
- virtual sound sources can even be positioned between the real speaker array and the listener.
- wave field synthesis is based on the principle of Huygens, according to which wavefronts can be formed and built up by superimposing elementary waves. After mathematically exact theoretical description, infinitely many sources in infinitely small distance would have to be used for the generation of the elementary waves. Practically, however, many speakers are finally used at a finite distance from each other. Each of these speakers is driven according to the WFS principle with an audio signal from a virtual source having a particular delay and a certain level. Levels and delays are usually different for all speakers.
- a wave field synthesis system operates on the Huygens principle and reconstructs a given waveform, for example, of a virtual source located at a certain distance from a listener by a plurality of single waves.
- the wave-field synthesis algorithm thus obtains information about the actual position of a single loudspeaker from the loudspeaker array and then calculates a component signal for this single loudspeaker that this loudspeaker ultimately has to emit, so that the listener can superimpose the loudspeaker signal from one loudspeaker to the loudspeaker signals from the other active loudspeaker Speaker performs a reconstruction in that the listener has the impression that he is not "sonicated" by many individual speakers, but only from a single speaker at the position of the virtual source.
- each virtual source For multiple virtual sources in a wave-field synthesis setting, the contribution from each virtual source is for each speaker, that is, the component signal of the first virtual source for the first loudspeaker, the second virtual source for the first loudspeaker, etc., to then add up the component signals to finally obtain the actual loudspeaker signal.
- the component signal of the first virtual source for the first loudspeaker For multiple virtual sources in a wave-field synthesis setting, the contribution from each virtual source is for each speaker, that is, the component signal of the first virtual source for the first loudspeaker, the second virtual source for the first loudspeaker, etc., to then add up the component signals to finally obtain the actual loudspeaker signal.
- superimposing the loudspeaker signals of all the active loudspeakers on the listener would mean that the listener does not feel that he is being sonicated by a large array of loudspeakers, but that the sound he hears is merely from three sound sources positioned at specific positions, which are equal to the virtual sources.
- the calculation of the component signals usually takes place in practice by applying a delay and a scaling factor to the audio source assigned to a virtual source, depending on the position of the virtual source and the position of the loudspeaker, at a specific point in time in order to obtain a delayed and / or scaled audio signal to obtain a virtual source that directly represents the loudspeaker signal when there is only one virtual source, or that after addition of other component signals for the considered loudspeaker from other virtual sources then contributes to the loudspeaker signal for the considered loudspeaker.
- Typical wave field synthesis algorithms operate regardless of how many speakers are present in the speaker array.
- the underlying theory of Wave Field Synthesis is that any sound field can be accurately reconstructed by an infinite number of individual speakers, with the individual individual speakers arranged infinitely close to each other. In practice, however, neither the infinitely high number nor the infinitely close arrangement can be realized. Instead, there are a limited number of speakers, which are also arranged at certain predetermined distances from each other. Thus, in real systems, only an approximation to the actual waveform that would occur if the virtual source were actually present, would be a real source.
- Wave field synthesis devices are also capable of replicating several different types of sources.
- a prominent source form is the point source, where the level decreases proportionally 1 / r, where r is the distance between a listener and the position of the virtual source.
- Another source form is a source that emits plane waves.
- the level remains constant regardless of the distance to the listener, since plane waves can be generated by point sources, which are arranged at an infinite distance.
- the object of the present invention is therefore to provide improved concepts for devices, methods and electroacoustic systems for reverberation time extension.
- the object of the present invention is achieved by a device according to claim 1, a method according to claim 11, a computer program according to claim 12, an electroacoustic system according to claim 13 and a method according to claim 14.
- the invention provides a device for reverberation time extension.
- the device comprises a module for calculating wave field synthesis information and a signal processor for generating a plurality of audio output signals for a plurality of loudspeakers based on a plurality of audio input signals, and based on the wave field synthesis information, wherein the audio signals have been picked up by a plurality of microphones.
- the device comprises an operating unit for determining a virtual position of one or more virtual walls.
- the wave field synthesis information calculation module is configured to calculate the wave field synthesis information based on the virtual position of the one or more virtual walls.
- the virtual position can be set by the operating unit for at least one of the virtual walls.
- the acoustic amplification is also achieved by a regenerative effect, which consists in that the output via speakers generated audio output signals are picked up again by the microphones and thus enter into the generation of the audio output signals at a later date.
- an apparatus and method for generating an acoustic spatial magnification wherein distributed microphones detect relevant sound sources and the acoustic environment and reproduce this with respect to fixed or dynamic virtual source positions via a wave field synthesis system.
- the invention is based on the concept that the virtual sources are generated in an algorithm based on wave field synthesis.
- the invention describes a method in which by means of distributed microphones in the room to be sounded, the acoustics of the room with the sources to be amplified is detected and AD converter is fed to a processing system.
- the processing system may consist of software in which the signal is first processed via filters, and then processed in a wave field synthesis algorithm to an object-based sound source, which in turn is processed via filters, to then be played over a wave field synthesis system. Due to the possibilities of wave field synthesis, the acquired room signals can now be positioned as desired and can be moved as "virtual walls" as desired. As a result, individual room geometries can be generated.
- the recorded space signals are typically represented as plane waves and thus correspond to the acoustic effect of a wall. Not only can this virtual wall be moved, it can also be changed in angle, directly affecting the reflection patterns of the sound sources.
- the wave field synthesis information calculation module is configured to calculate delay values and amplitude factor values as wave field synthesis information.
- the delay value indicates the delay by which one of the audio input signals is delayed at one of the speakers.
- the amplitude factor value indicates by what factor the amplitude of one of the audio input signals is modified to obtain a modified signal output at one of the speakers.
- the wave field synthesis information calculation module may be configured to calculate a delay value and an amplitude factor value for each speaker-virtual wall pair at a time, wherein a speaker-virtual wall pair, a pair of one of the speakers and one of the virtual walls is.
- the wave field synthesis information calculation module is configured to apply the delay value and the amplitude factor value to a speaker-virtual wall pair based on the distance from the speaker and the virtual wall of the speaker-virtual wall pair to calculate.
- the wave field synthesis information calculation module may be configured to set the delay value of a speaker virtual wall pair the larger the distance between the speaker and the virtual wall.
- the wave field synthesis information calculation module may be configured to set the amplitude factor value of a speaker-virtual wall pair the smaller the distance between the speaker and the virtual wall.
- the operating unit is designed so that at least one of the virtual walls is displaceable from a first virtual position to a second virtual position, so that the virtual wall is displaceable in any desired parallel position relative to its first position. Furthermore, the operating unit is designed such that at least one of the virtual walls is displaceable from a first virtual position to a second virtual position, so that the virtual wall can be displaced as desired in relation to its first position.
- the operating unit is designed so that the virtual position can be set by the operating unit for all of the virtual walls.
- the operating unit can be designed such that each of the virtual walls can be displaced from a first virtual position to a second virtual position. so that each virtual wall is arbitrarily parallel and rotatable relative to its first position.
- the reverberation time extension apparatus may include a parametric filter for filtering resonant frequencies.
- an electro-acoustic system for Reverberation time extension provided that a plurality of microphones, a reverberation time extension device according to one of the embodiments described above and a loudspeaker array comprises a plurality of speakers.
- the plurality of microphones are configured to generate a plurality of audio input signals fed to the reverberation time extension apparatus, and wherein the plurality of loudspeakers of the loudspeaker array are adapted to receive and feed the audio output signals from the reverberation time extension apparatus Play audio output signals.
- Fig. 1 shows a device for reverberation time extension according to one embodiment.
- the apparatus includes a wave field synthesis information calculation module 110. Further, the apparatus includes a signal processor 120 for generating a plurality of audio output signals y 1 , y 2 ,..., Y n for a plurality of loud speakers (not shown) based on a plurality of audio input signals s 1 , s 2 , ..., s n picked up by a plurality of microphones (not shown) and based on the wave field synthesis information. Furthermore, the device comprises an operating unit 130 for determining a virtual position of one or more virtual walls.
- the wave field synthesis information calculation module 110 is configured to calculate the wave field synthesis information WS inf based on the virtual position of the one or more virtual walls. In this case, its virtual position can be set by the operating unit for at least one of the virtual walls.
- the wave field synthesis information calculation module 110 and the signal processor 120 may be implemented in one module (a wave field synthesis module).
- Fig. 2 an embodiment of the interaction of the module for calculating wave field synthesis information and the signal processor set forth.
- the wave field synthesis information calculation module 210 and the signal processor 220 are shown by dashed lines.
- the wave field synthesis information calculation module 210 and the signal processor 220 have a highly parallel construction in that, starting from the audio signal supplied to the signal processor for each virtual wall (a virtual source) and from the position information for the corresponding virtual wall (virtual source), has received the module for calculating wave field synthesis information 210 from a control unit, first delay information (delay information) V i and amplitude factors (scaling factors) SF i are calculated, which depends on the position information and the position of the currently considered speaker, for. B. the speaker with the ordinal number j, so LS j depend.
- a discrete value AW i (t A ) for the Computed signal K ij calculated in a final received speaker signal This is done by means 310, 312, 314, 316, as they are in Fig. 2 are shown schematically. Fig. 2 also shows a kind of "flash photography" at time t A for the individual component signals. The individual component signals are then summed by a summer 320 in nodes to determine the discrete value for the current time t A of the speaker signal for the loudspeaker j, which can then be supplied to the speaker for the output.
- Fig. 2 Like it out Fig. 2 is apparent, first for each virtual source, individually calculate a value valid on the basis of delay information (delay) and scaling with an amplitude factor (scaling factor) at a current time, after which all the component signals for a loudspeaker are summed due to the different virtual walls (virtual sources). For example, if only one virtual source were present, then the summer would be omitted, and that at the output of the summer in Fig. 2 applied signal would z. For example, correspond to the signal output from the device 310 when the virtual source 1 is the only virtual source.
- delay information delay information
- scaling factor scaling factor
- Fig. 3 illustrates an electroacoustic WFS system for reverberation time extension according to an embodiment.
- Fig. 3 are installed in a room evenly four microphones 350 1m suspended from the ceiling.
- the microphone signals are processed in a WFS algorithm 360 to a virtual source, which is reproduced as a plane wave via a WFS sound system 370 in the same room.
- the WFS system includes a user interface for moving the 4 microphone sources. With this control unit, the 4 microphone signals are detected and pulled outwards. The result is an acoustic enlargement of the room.
- the wave field synthesis module 360 includes a wave field synthesis information calculation module according to the embodiment of FIG Fig. 1 and a signal processor according to the embodiment of FIG Fig. 1 ,
- the operating unit 375 is an operating unit according to the embodiment of FIG Fig. 1 ,
- unit 355 sets in Fig. 3 a filter which serves to filter resonance frequencies.
- a sound wave output at one of the speakers 370 is resumed by the microphones 350 and re-considered in the generation of the later audio signal output via the speakers.
- filter 355 can be used to suppress these resonances.
- filter 365 may be a conventional filter used, for example, to adapt the speakers.
- Fig. 4 illustrates another embodiment of an electroacoustic WFS system.
- the signals of the ceiling microphones are processed in a central processing unit and processed after filtering in a matrix to virtual sources, which is played back after level adjustment, control of the spatial proportion and speaker filtering as virtual sound sources via a WFS array and evenly distributed ceiling speakers.
- Microphones feed 411, 412 audio input signals into Microphone Preamplifier 416, 417.
- the microphone 411 is a microphone that is close to a sound source such as a lectern.
- the microphone 412 is a room microphone located in the room but farther from the sound source than the microphone 411. Typically, multiple room microphones and / or multiple microphones are used close to the sound source.
- Microphone preamplifiers 416, 417 amplify the audio input signals received from the microphones 411, 412 to obtain preamplified audio input signals.
- the microphone preamplifiers 416, 417 may be conventional microphone preamplifiers.
- the pre-amplified audio input signals are fed to an analog-to-digital converter 420, which converts the audio input signals, which are initially in analog form, into digital audio signals.
- the analog-to-digital converter 420 may be a conventional analog-to-digital converter.
- the analog-to-digital converter 420 feeds the digital audio signals into the absorption filter 425.
- Absorption filter 425 performs filtering to match the wall material.
- absorption filter 425 filters such that when highly reflective walls are to be replicated, the digital audio signals pass absorption filter 425 almost unfiltered.
- absorption filter 425 in one embodiment filters the digital audio signals to a great extent.
- Filter 430 is a filter for feedback compensation and sound adjustment. If a signal is reproduced by a loudspeaker, the sound waves of this signal are in turn detected by the microphone and this leads to a feedback. In one embodiment, filter 430 may be used to compensate for this feedback in whole or in part. In addition, Filter 430 can be used for sound adjustment. In one embodiment, the feedback compensation and / or the sound adjustment may be performed in a conventional manner.
- the system includes in Fig. 4 a central operating unit 435 and a module for calculating wave field synthesis information 440.
- the central operating unit 435 of the operating unit in Fig. 1 correspond.
- the central operating unit in FIG. 4 can be equipped with a GUI (Graphical User Interface).
- the wave field synthesis information calculation module 440 may be adapted to the wave field synthesis information calculation module Fig. 1 correspond.
- the wave field synthesis information calculation module 440 passes the calculated wave field synthesis parameters to module 445.
- These wave field synthesis parameters may be e.g. to act on delay values and amplitude values, such as amplitude factor values.
- Module 445 builds a Delay Amplitude Matrix from the values passed by Module 440.
- the delay amplitude matrix may include a delay value and an amplitude factor value for a particular point in time for each speaker-virtual wall pair.
- Module 445 performs audio scaling based on the wave field synthesis parameters obtained from the wave field synthesis information calculation module 440. For example, if a delay value and an amplitude factor value were obtained for a loudspeaker-virtual wall pair, the signal originating from the virtual wall (eg, apparently reflected from the virtual wall) will be the received delay value is delayed, and the amplitude factor value obtained from module 440 is modified to the amplitude of the signal to be output by the amplitude factor value, for example by multiplying the amplitude factor value by the amplitude of the signal to be output.
- filter 450 filters the module 445 modified audio signals to achieve speaker matching.
- the audio signals are modified to adjust the overall volume. This can be done in the usual way.
- the ratio of volume proportion to original signal is adjusted. For example, in one embodiment, the ratio of audio signals generated from audio signals of room microphones to audio signals generated from audio signals of microphones near the lectern can be adjusted, for example, by adjusting the amplitudes of the respective signals.
- the modified digital audio signals are then fed to a digital-to-analog converter 465 which converts the modified digital audio signals to analog audio output signals.
- the analog audio output signals are then amplified by power amplifiers 471, 472 and output from loudspeakers 481, 482.
- the audio signals are output from either WFS speakers 481 or ceiling speakers 482. It is understood that in a real system a variety of WFS speakers and / or ceiling speakers can be used.
- Fig. 5 shows a medium conference room (5mx18x15m) equipped with 5 ceiling microphones 511, 512, 513, 514, 515, 40 ceiling loudspeakers and a circulating horizontal band of conventional loudspeakers in a reduced WFS arrangement 530.
- the signals of the ceiling microphones 511, 512, 513, 514, 515 are processed in a central processing unit and processed after filtering in a matrix to virtual sources, which after level adjustment, control of the spatial proportion and speaker filtering as virtual sound sources 521, 522, 523, 524, 525 via a WFS array and evenly distributed ceiling speakers is played again.
- the structure shows FIG. 4 ,
- the microphone signals are represented by the respectively opposite virtual sources in order to avoid feedback.
- the input branch of the matrix also contains a filter unit which takes into account different spatial materials in order to incorporate various absorption and reflection parameters into the space to be reverberated.
- the recorded microphone signals are, as described, reproduced in freely positionable sources and acted upon by the existing room characteristics again captured by the microphone, resulting in a regeneration of the room acoustics.
- Fig. 6 illustrates a basic concept of certain embodiments. Shown is a speaker array that, as in Fig. 6 12 speakers 611, 612, 613. In actual embodiments, the number of loudspeakers will often be significantly larger and include, for example, 60, 100, 200, 300 or more loudspeakers. Also shown are four virtual walls 621, 622, 623, 624.
- the speaker 611 and the virtual wall 621 will be considered in more detail. These form a speaker-virtual wall pair (611, 621). Also, any other combination of one of the speakers and one of the virtual walls forms a speaker-virtual wall pair. The distance between the speaker and the virtual wall is indicated by an arrow d. In Fig. 6 In addition, a plurality of microphones 631, 632, 633 is provided. For the sake of simplicity, it is assumed that a microphone 631 generated by recording sound waves, an audio signal to be reproduced via the speaker 611. In this case, the signal reproduced by the loudspeaker 611 should correspond to a reflection of the sound waves recorded by the microphone 631 on the virtual wall 621.
- the signal picked up by the microphone can only be reproduced with a time delay by the loudspeaker 611, which depends on the distance between the loudspeaker and the virtual wall: the greater the distance between virtual wall 621 and loudspeaker 611, the greater the temporal Delay, ie the delay value with which the signal picked up by the microphone 631 is to be reproduced on the loudspeaker 611.
- Fig. 6 shows by the dashed line 629 a displacement of the virtual wall 621, wherein the distance of the virtual wall of the speaker 611 increases from d to 2d. The delay value will increase accordingly.
- delay d + c * p 1 , where d is the distance between the speaker and the virtual wall of the speaker-virtual wall pair, c is a constant value, and p 1 is a proportionality constant greater than 0. The greater the distance between the speaker and the virtual wall, the greater the delay value becomes.
- the amplitude factor is the factor with which the amplitude of one of the output signals is to be modified in order to obtain a modified signal to be output at one of the loudspeakers.
- an increase in the delay value can bring about a reverberation time extension.
- Fig. 7 shows according to the invention that the current position 729 of the virtual wall is changed such that the current position 729 of the virtual wall has been rotatably changed with respect to its old position 721.
- the distance of the old position of the virtual wall of speaker 711 is indicated by arrow e
- the distance of the new position of the virtual wall from the speaker 711 is shown by arrow f.
- a computer program or signal according to the invention may be stored on a digital storage medium or may be transmitted on a transmission medium, e.g. a wireless transmission medium or a wired transmission medium, e.g. the internet.
- a transmission medium e.g. a wireless transmission medium or a wired transmission medium, e.g. the internet.
- embodiments of the invention may be implemented in hardware or in software.
- the implementation may be done using a digital storage medium, such as a digital storage medium. a floppy disk, a DVD, a CD, a ROM, a PROM, an EPROM, an EEPROM, or a FLASH memory which stores electronically readable control signals that cooperate (or are able to work together) with a programmable computer system so that the appropriate procedure is carried out.
- a digital storage medium such as a digital storage medium. a floppy disk, a DVD, a CD, a ROM, a PROM, an EPROM, an EEPROM, or a FLASH memory which stores electronically readable control signals that cooperate (or are able to work together) with a programmable computer system so that the appropriate procedure is carried out.
- Some embodiments according to the invention comprise a non-transitory data carrier having electronically readable control signals capable of cooperating with a programmable computer system to perform one of the methods described herein.
- embodiments of the present invention may be implemented as a computer program product having program code, wherein the program code is operative to perform one of the methods when the computer program product is executed on a computer.
- the program code may e.g. be stored on a machine-readable carrier.
- inventions include the computer program for executing one of the methods described herein stored on a machine readable carrier.
- an embodiment of the method according to the invention is therefore a computer program with a program code for carrying out one of the methods described herein, when the computer program is running on a computer.
- a further embodiment of the inventive method is therefore a data carrier (or a digital storage medium or a computer-readable medium) recorded thereon the computer program for carrying out one of the methods described herein.
- a further embodiment of the method according to the invention is therefore a data stream or a series of signals representing the computer program for carrying out one of the methods described herein.
- the data stream or signal series may e.g. configured to be transmitted over a data communication link, e.g. over the internet.
- processing means e.g. a computer, or programmable logic device configured or adapted to perform one of the methods described herein.
- Another embodiment includes a computer on which the computer program is installed to perform one of the methods described herein.
- a programmable logic device e.g., a field programmable gate array
- a field programmable gate array may cooperate with a microprocessor to perform any of the methods described herein. In general, the methods are preferably performed by any hardware device.
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Claims (14)
- Dispositif de prolongement d'un temps de réverbération, comportant:un module (110) destiné à calculer l'information de synthèse de champ d'onde,un processeur de signal (120) destiné à générer une pluralité de signaux de sortie audio pour une pluralité de haut-parleurs sur base d'une pluralité de signaux d'entrée audio qui sont enregistrés par une pluralité de microphones, et sur base de l'information de synthèse de champ d'onde, etune unité de commande (130) destinée à établir une position virtuelle d'une ou plusieurs parois virtuelles,dans lequel le module (110) destiné à calculer l'information de synthèse de champ d'onde est conçu pour calculer l'information de synthèse de champ d'onde sur base de la position virtuelle d'une ou plusieurs parois virtuelles, etdans lequel, pour au moins l'une des parois virtuelles, la position virtuelle peut être réglée par l'unité de commande (130),dans lequel l'unité de commande (130) est conçue pour qu'au moins l'une des parois virtuelles puisse être déplacée d'une première position virtuelle en une deuxième position virtuelle, de sorte que la paroi virtuelle puisse être déplacée à volonté par rotation par rapport à sa première position.
- Dispositif selon la revendication 1, dans lequel le module (110) destiné à calculer l'information de synthèse de champ d'onde est conçu pour calculer des valeurs de retard et des valeurs de facteur d'amplitude comme information de synthèse de champ d'onde, la valeur de retard indiquant le retard avec lequel l'un des signaux d'entrée audio est reproduit retardé à l'un des haut-parleurs, et le facteur d'amplitude indiquant le facteur par lequel l'amplitude de l'un des signaux d'entrée audio est modifié pour obtenir un signal modifié qui est sorti à l'un des haut-parleurs.
- Dispositif de la revendication 2, dans lequel le module (110) destiné à calculer l'information de synthèse de champ d'onde est conçu pour calculer, pour chaque paire de haut-parleur et paroi virtuelle pour un moment, une valeur de retard et une valeur de facteur d'amplitude, où une paire de haut-parleur et paroi virtuelle est une paire constituée d'un des haut-parleurs et d'une des parois virtuelles.
- Dispositif selon la revendication 3, dans lequel le module (110) destiné à calculer l'information de synthèse de champ d'onde est conçu pour calculer la valeur de retard et la valeur de facteur d'amplitude pour une paire de haut-parleur et paroi virtuelle sur base de la distance entre le haut-parleur et la paroi virtuelle de la paire de haut-parleur et paroi virtuelle.
- Dispositif selon la revendication 3 ou 4, dans lequel le module (110) destiné à calculer l'information de synthèse de champ d'onde est conçu pour fixer la valeur de retard d'une paire de haut-parleur et paroi virtuelle d'autant plus grande que la distance entre le haut-parleur et la paroi virtuelle est grande.
- Dispositif selon l'une des revendications 3 à 5, dans lequel le module (110) destiné à calculer l'information de synthèse de champ d'onde est conçu pour fixer la valeur d'amplitude d'une paire de haut-parleur et paroi virtuelle d'autant plus petite que la distance entre le haut-parleur et la paroi virtuelle est grande.
- Dispositif selon l'une des revendications précédentes, dans lequel l'unité de commande (130) est conçue de sorte qu'au moins l'une des parois virtuelles puisse être déplacée d'une première position virtuelle en une deuxième position virtuelle, de sorte que la paroi virtuelle puisse être déplacée de manière parallèle à volonté par rapport à sa première position.
- Dispositif selon l'une des revendications précédentes, dans lequel pour toutes les parois virtuelles, la position virtuelle est réglable par l'unité de commande (130).
- Dispositif selon l'une des revendications précédentes, dans lequel l'unité de commande (130) est conçue de sorte que chacune des parois virtuelles puisse être déplacée d'une première position virtuelle en une deuxième position virtuelle, de sorte que chaque paroi virtuelle puisse être déplacée de manière parallèle et rotative à volonté par rapport à sa première position.
- Dispositif selon l'une des revendications précédentes, dans lequel le dispositif comporte par ailleurs un filtre paramétrique destiné à filtrer des fréquences de résonance.
- Procédé de prolongement du temps de réverbération, comprenant le fait de:fixer une position virtuelle d'une ou plusieurs parois virtuelles,recevoir une pluralité de signaux d'entrée audio qui sont enregistrés d'une pluralité de microphones,calculer l'information de synthèse de champ d'onde, etgénérer une pluralité de signaux de sortie audio pour une pluralité de haut-parleurs sur base des signaux d'entrée audio et sur base de l'information de synthèse de champ d'onde,dans lequel l'information de synthèse de champ d'onde est calculée sur base de la position virtuelle des une ou plusieurs parois virtuelles, etdans lequel la position virtuelle est réglable pour au moins l'une des parois virtuelles,dans lequel au moins l'une des parois virtuelles est déplacée d'une première position virtuelle en une deuxième position virtuelle, de sorte que la paroi virtuelle soit déplacée par rotation par rapport à sa première position.
- Programme d'ordinateur avec un code de programme pour réaliser le procédé selon la revendication 11 lorsque le programme d'ordinateur est exécuté sur un ordinateur.
- Système électro-acoustique de prolongement de temps de réverbération, comportant:une pluralité de microphones (350; 411, 412),un dispositif de prolongement du temps de réverbération selon l'une des revendications 1 à 10, etun réseau de haut-parleurs comprenant une pluralité de haut-parleurs (370; 481, 482),dans lequel la pluralité de microphones (350, 411, 412) sont conçus pour générer une pluralité de signaux d'entrée audio qui sont alimentés dans le dispositif de prolongement du temps de réverbération, et dans lequel la pluralité de haut-parleurs (370; 481, 482) du réseau de haut-parleurs sont conçus pour recevoir les signaux de sortie audio alimentés par le dispositif de prolongement du temps de réverbération, et pour reproduire les signaux de sortie audio alimentés.
- Procédé de prolongement du temps de réverbération à l'aide d'un système électro-acoustique, comprenant le fait de:enregistrer une pluralité de signaux d'entrée audio d'une pluralité de microphones,mettre en oeuvre le procédé de prolongement du temps de réverbération selon la revendication 11 pour générer une pluralité de signaux de sortie audio, l'étape consistant à recevoir la pluralité de signaux d'entrée audio comprenant le fait que sont reçus la pluralité signaux d'entrée audio qui ont été enregistrés par la pluralité de microphones, etsortir la pluralité de signaux de sortie audio par l'intermédiaire d'un réseau de haut-parleurs comprenant une pluralité de haut-parleurs.
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US201161531899P | 2011-09-07 | 2011-09-07 | |
DE102011082310A DE102011082310A1 (de) | 2011-09-07 | 2011-09-07 | Vorrichtung, Verfahren und elektroakustisches System zur Nachhallzeitverlängerung |
PCT/EP2012/066392 WO2013034444A1 (fr) | 2011-09-07 | 2012-08-23 | Dispositif, procédé et système électroacoustique de prolongement du temps de réverbération |
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EP2754151A1 EP2754151A1 (fr) | 2014-07-16 |
EP2754151B1 EP2754151B1 (fr) | 2016-01-06 |
EP2754151B2 true EP2754151B2 (fr) | 2018-10-31 |
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EP (1) | EP2754151B2 (fr) |
JP (1) | JP5995973B2 (fr) |
CN (1) | CN103907151B (fr) |
DE (1) | DE102011082310A1 (fr) |
WO (1) | WO2013034444A1 (fr) |
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EP2777301B1 (fr) | 2011-11-10 | 2015-08-12 | SonicEmotion AG | Procédé d'implémentations pratiques de reproduction de champs sonores basé sur des intégrales de surface en trois dimensions |
US20160173808A1 (en) * | 2014-12-16 | 2016-06-16 | Psyx Research, Inc. | System and method for level control at a receiver |
CN107281753B (zh) * | 2017-06-21 | 2020-10-23 | 网易(杭州)网络有限公司 | 场景音效混响控制方法及装置、存储介质及电子设备 |
CN109195062B (zh) * | 2018-09-21 | 2020-10-02 | 歌尔科技有限公司 | 一种扩大音频设备的声场的方法、***及音频设备 |
WO2021117576A1 (fr) * | 2019-12-13 | 2021-06-17 | ソニーグループ株式会社 | Dispositif de traitement de signal, procédé de traitement de signal et programme |
CN113965842A (zh) * | 2021-12-01 | 2022-01-21 | 费迪曼逊多媒体科技(上海)有限公司 | 一种基于wfs波场合成技术的可变声学家庭影院音响*** |
DE102022129642A1 (de) * | 2022-11-09 | 2024-05-16 | Holoplot Gmbh | Verfahren zur richtungsabhängigen Korrektur des Frequenzganges von Schallwellenfronten |
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JP2569872B2 (ja) * | 1990-03-02 | 1997-01-08 | ヤマハ株式会社 | 音場制御装置 |
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AU8006094A (en) | 1993-10-15 | 1995-05-04 | Industrial Research Limited | Improvements in reverberators for use in wide band assisted reverberation systems |
JP3722335B2 (ja) | 1998-02-17 | 2005-11-30 | ヤマハ株式会社 | 残響付加装置 |
DE10254404B4 (de) | 2002-11-21 | 2004-11-18 | Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. | Audiowiedergabesystem und Verfahren zum Wiedergeben eines Audiosignals |
DE10254470B4 (de) * | 2002-11-21 | 2006-01-26 | Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. | Vorrichtung und Verfahren zum Bestimmen einer Impulsantwort und Vorrichtung und Verfahren zum Vorführen eines Audiostücks |
DE10321986B4 (de) | 2003-05-15 | 2005-07-14 | Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. | Vorrichtung und Verfahren zum Pegel-Korrigieren in einem Wellenfeldsynthesesystem |
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DE102004002532A1 (de) | 2004-01-17 | 2005-09-22 | Helmut Oellers | Frontalmatrix-Wellenfeldsynthese (FMWFS) |
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DE102005008366A1 (de) | 2005-02-23 | 2006-08-24 | Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. | Vorrichtung und Verfahren zum Ansteuern einer Wellenfeldsynthese-Renderer-Einrichtung mit Audioobjekten |
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DE102005027978A1 (de) | 2005-06-16 | 2006-12-28 | Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. | Vorrichtung und Verfahren zum Erzeugen eines Lautsprechersignals aufgrund einer zufällig auftretenden Audioquelle |
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KR20100116223A (ko) * | 2008-03-20 | 2010-10-29 | 프라운호퍼 게젤샤프트 쭈르 푀르데룽 데어 안겐반텐 포르슝 에. 베. | 음향 디스플레이 장치 및 방법 |
JP5199915B2 (ja) * | 2009-02-18 | 2013-05-15 | キヤノン株式会社 | 音場補正方法及び音場補正装置 |
JP4883197B2 (ja) * | 2010-02-15 | 2012-02-22 | ソニー株式会社 | 音声信号処理方法、音場再現システム |
-
2011
- 2011-09-07 DE DE102011082310A patent/DE102011082310A1/de not_active Ceased
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2012
- 2012-08-23 JP JP2014528927A patent/JP5995973B2/ja active Active
- 2012-08-23 WO PCT/EP2012/066392 patent/WO2013034444A1/fr active Application Filing
- 2012-08-23 CN CN201280054539.4A patent/CN103907151B/zh active Active
- 2012-08-23 EP EP12756411.0A patent/EP2754151B2/fr active Active
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2014
- 2014-03-06 US US14/199,362 patent/US9355632B2/en active Active
Also Published As
Publication number | Publication date |
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JP5995973B2 (ja) | 2016-09-21 |
CN103907151B (zh) | 2016-08-24 |
EP2754151B1 (fr) | 2016-01-06 |
WO2013034444A1 (fr) | 2013-03-14 |
US9355632B2 (en) | 2016-05-31 |
CN103907151A (zh) | 2014-07-02 |
DE102011082310A1 (de) | 2013-03-07 |
JP2014529251A (ja) | 2014-10-30 |
US20140185817A1 (en) | 2014-07-03 |
EP2754151A1 (fr) | 2014-07-16 |
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