EP1900250B1 - Procede electro-acoustique - Google Patents

Procede electro-acoustique Download PDF

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Publication number
EP1900250B1
EP1900250B1 EP06777451A EP06777451A EP1900250B1 EP 1900250 B1 EP1900250 B1 EP 1900250B1 EP 06777451 A EP06777451 A EP 06777451A EP 06777451 A EP06777451 A EP 06777451A EP 1900250 B1 EP1900250 B1 EP 1900250B1
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Prior art keywords
sound
impulse response
signals
room
optimised
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German (de)
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EP1900250A2 (fr
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Gunter Engel
Karlheinz Müller
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Mueller Bbm GmbH
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Mueller Bbm GmbH
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    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R27/00—Public address systems
    • 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
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R2227/00—Details of public address [PA] systems covered by H04R27/00 but not provided for in any of its subgroups
    • H04R2227/007—Electronic adaptation of audio signals to reverberation of the listening space for PA

Definitions

  • the present invention relates to a method for improving the perceived acoustics of a room, in particular for improving the room acoustics of an event hall, an open venue or the like by means of electro-acoustic measures.
  • the room acoustic conditions usually only meet very limited requirements of different types of events.
  • the room acoustic requirements for language events, music theater or music events with different instrumental composition differ significantly.
  • Speech events such as speech theater or lectures require strong early reflections and sufficient signal volume to ensure good intelligibility.
  • a reverberation time for such rooms for mainly language events should be between 0.5 and 1.5 s.
  • musical events however, a stronger reverberation of the room depending on volume, style of music and occupation of the orchestra between 1.0 and 2.5 s in classical music and even 2 to 10 s in sacred music is desired.
  • lateral reflections to increase the spatiality and the perceived sound source size should be aimed at.
  • the natural room acoustic conditions in function rooms are tailored to a specific type of event, such as a philharmonic orchestra, opera halls or lecture theaters, so that the acoustic results for different event types are often unsatisfactory.
  • a specific type of event such as a philharmonic orchestra, opera halls or lecture theaters
  • an adaptation of the room acoustic conditions by variable absorber surfaces, adjustable reflectors or additional Hall chambers done, but this requires a high technical and financial effort.
  • electroacoustic systems have been developed with which additional reflections and reverberations can be recorded in different areas of space.
  • a particular difficulty is then a sound discoloring feedback of the rehearsed To prevent signals in recording microphones. Often, this is done by using microphones and loudspeakers with a pronounced directivity and by using dynamic filters to suppress any feedback frequencies.
  • a homogeneous recording of the sound sources in the entire stage area and of high quality is required for systems for electronic room acoustics influencing.
  • the microphone arrangement should not be changed as far as possible in the case of different types of events. Highly directional microphones for suppressing feedback are therefore rather unsuitable.
  • stage situation is very variable at opera performances or in multi-purpose halls, whereby the natural room-acoustic conditions can also change during the performance. A repositioning of recording microphones is then hardly possible.
  • a plurality of optimized sound signals are generated from a sound signal of the corresponding sound source in the room, such as music or speech, by convolution with predetermined impulse response functions and reproduced by a plurality of speakers distributed in the room.
  • the feedback security is achieved in that the corresponding emission locations undergo a periodic change in switching time intervals.
  • the inventive method leads to a particularly pleasant spatial sound.
  • additional reflections or reverberation often have to be recorded just in the vicinity of the stage or the sound source in order to achieve the most pleasant possible room acoustics. In prior art systems, this has often been problematic since it can easily cause feedback.
  • the recording of additional reverberation in the vicinity of the stage is also made possible.
  • At least three different optimized sound signals are generated, and at least three reproduction means are provided.
  • the switching time interval is chosen so short that no sound-discoloring feedback is produced between a respective reproducing means and the recording means.
  • a sound discoloring usually unwanted level increases limited frequency ranges are felt.
  • a clearly perceptible whistling for example, such increases in the signal level are disturbing even by local frequencies.
  • such possible color discolorations are counteracted by the switching.
  • the switching time interval depends on the distance between the corresponding recording medium or recording microphone and the speakers arranged in the hall for reproducing the optimized sound signals.
  • a preferred switching time interval is approximately in a range between 100 ms and 500 ms.
  • the switching time interval is changed over time.
  • Different interval durations can be set, for example, depending on the type of sound source and event.
  • the interval duration can also be selected randomly, so that the optimized sound signals are fed completely stochastically to the speakers.
  • the respective first sound signal optimized in the respective second sound signal is soft-blended.
  • Such a smooth transition results in a particularly homogeneous sound carpet, which is perceived as pleasant in certain types of sound sources.
  • the optimized sound signals are preferably distributed in a periodic order or randomly stochastich distributed to the reproducing means.
  • a number of optimized sound signals is generated in step b), which is greater than the number of playback means. Then, at least one of the generated optimized sound signals is generated during a respective switching interval not guided to a playback means. In this alternative method, therefore, one of the optimized sound signals is not reproduced, whereby a certain randomization of the reproduction of all optimized sound signals is achieved.
  • the given impulse response functions are uncorrelated with each other.
  • the predetermined impulse response functions are generated for multiple locations in the room. Because the natural impulse response functions are first measured for the adjustment or installation of the method according to the invention in the corresponding room, certain proportions can be added to determine the predetermined impulse response functions in order to achieve the most ideal impulse response function possible. By using impulse response functions measured in the room and by folding the sound signals recorded by the sound source with these modified original impulse response functions, a particularly natural sound impression is achieved according to the invention.
  • the predetermined impulse response functions can be generated such that early reflection components in a respective impulse response function substantially correspond to an impulse response obtained by a microphone recording with a high direct sound component and strong early reflections.
  • natural microphone recordings are used to alter the original impulse response function measured in the room such that additional early reflections enhancing the direct sound component are perceived become.
  • the predetermined impulse response functions are preferably generated such that reverberation portions substantially correspond to microphone recordings having a high diffuse sound.
  • impulse response functions that appear as natural as possible for convolution, which consist of impulse response functions of the room and have additional components for direct sound and diffuse sound. It is equally possible to remove disturbing components in a measured impulse response function.
  • the realistic acoustic properties corresponding to the room are first of all determined and reworked or "improved". In this case, therefore, preferably not completely synthetically generated Hall signals are used but "natural" Hall of the room prepared.
  • the optimized sound signals during reproduction are subjected to a periodic level change.
  • This periodic level change further reduces the risk of feedback and creates an improved sense of space in a listener.
  • the time constant of the periodic level change is then selected as a function of a time constant for a feedback setup in the room.
  • the optimized sound signals are delayed such that the reproduced optimized sound signals and the sound signals of the sound source arrive at at least one location of the room substantially simultaneously and preferably the sound signals of the sound source arrive earlier.
  • the optimized sound signals are delayed such that a direct sound of the sound sources at each location of the room always arrives in time before a reflection component of a reproduced optimized sound signal.
  • a plurality of individual sound signals of the sound source are recorded via a plurality of recording means and the corresponding individual sound signals are mixed with one another to form a recorded sound signal for further processing.
  • the individual sound signals are delayed depending on the positions of the recording means.
  • the single-tone signals or the recorded sound signals are filtered to reduce higher-frequency components.
  • the absorption property of the transmission medium for the sound in the room can be simulated and taken into account.
  • the predetermined impulse response functions are preferably stored in the sound system according to the invention in a memory coupled to the convolution device. Furthermore, it is preferable for the switching device to be followed by a level adjusting device for periodically changing the levels of the optimized sound signals and / or a delay device for delaying the optimized sound signals.
  • the speakers in the room are advantageously arranged such that preferably at least three speakers are perceptible at a particular location by a listener in the room.
  • an improvement of the sound properties is already possible by the electroacoustic measures according to the invention.
  • Perceptible in this case does not necessarily mean that an amplification of the original sound signal of the sound source is made and a listener hears the appropriately rehearsed signals, but that the sound enhancement or the improvement of the acoustics of the room is perceptible.
  • a particularly preferred embodiment of the sound system according to the invention provides a plurality of signal branches, each with at least one recorded sound signal, a convolution device, a switching device and a group of speakers, wherein the corresponding predetermined impulse response functions are selected in dependence on the positions of the speakers.
  • the invention further comprises an event hall equipped with a sound system according to the invention.
  • the Fig. 1 shows an inventive sound system 1 for a room 2, for example, a concert or performance hall.
  • a stage area 3 is provided, which serves here schematically as a sound source.
  • a sound signal E1 (t) is recorded via a microphone 4.
  • the recorded sound signal E1 (t) is fed to a convolver 5, which may be implemented, for example, as a digital signal processor or PC.
  • the folding device 5 is coupled to a memory 6 which holds stored impulse response functions stored.
  • the three optimized sound signals L1, L2, L3 are supplied to a switching device 7, which is coupled to the folding device 5.
  • the switching device has a first, second and third input 8, 9 and 10 and three outputs 11, 12 and 13.
  • the switching device 7 switches the optimized sound signals L1, L2, L3 in such a way to their outputs 11, 12 and 13 that, for example during a first Umschaltzeitvalvall the length .DELTA.t the signal L1 can be tapped as a loudspeaker signal N1 at the first output 11, the second optimized sound signal L2 at the second output 12 as the second loudspeaker signal N2 can be tapped off and the third optimized sound signal L3 at the third output 13 as the third Speaker signal N3 can be tapped.
  • L1 is then fed to the output 12, L2 to the output 13 and L3 to the output 11.
  • a listener in the room 2 for example at the place 17, on the one hand perceives the direct sound from the stage 3 and on the other hand reverberation parts which are emitted by the system 1 according to the invention through the loudspeakers 14, 15, 16.
  • the subjective sound impression for a listener results essentially from the original acoustic properties of the room 2 and the predetermined impulse responses with which the recorded signal E1 (t) is folded.
  • the acoustics of a room can be characterized by its impulse response or its corresponding transfer functions.
  • the corresponding impulse response function designates the sound pressure profile at a specific location of the room as a function of time when a delta-shaped acoustic wave is emitted at another location in the room.
  • An exemplary impulse response gi (t) is in the Fig. 2 shown.
  • a short pulse such as a shot or clapping, caused and recorded the corresponding sound pressure or the energy density of the sound signal elsewhere.
  • the solid curve A1 can, for example, the impulse response function of in Fig. 1 represented space 2 for a transmitted pulse on the stage 3 and the received sound pressure at the position 20 correspond.
  • the first early components correspond to a direct sound DS, which propagates without any detours via reflections on any surfaces to the receiver or listener.
  • the parts of the impulse response function gi (t) present in a range up to approximately 80 ms after the direct sound DS are usually referred to. Hall components at longer times are perceived as diffuse DF.
  • rooms which have an impulse response function which corresponds approximately to a "fir-tree" and an envelope, such as, for example, the curve A2 are perceived as particularly pleasant.
  • the predetermined and stored impulse response functions which are used in the convolution device 5 can now be generated in such a way that, for a harmonic sound impression, missing reverberation parts, for example the dashed line A3 in FIG Fig. 2 , which are added to the measured impulse response function.
  • This can be done, for example, by recording impulse response functions in rooms with particularly good acoustics, and then inserting such reverberation units into a measured impulse response of the room to be improved.
  • disturbing components in a measured impulse response function can be removed and the predetermined impulse response functions generated in this way can be used as the basis for the convolution in the convolver 5.
  • the sound characteristics of the second space can be impressed on the sound signals.
  • the folding takes place as described in equation (1).
  • feedback can occur and a listener would also be able to locate the additionally rehearsed reverb via the speakers, which would normally be annoying.
  • the folded or optimized signals L1, L2, L3 are therefore replaced by the switching device 7.
  • predetermined impulse response functions for convolution for example impulse response functions recorded in the hall at particularly favorable listener positions come into consideration, such as, for example, in the first third of a recitaled concert hall. Then, at adjacent positions 17, 18, 19, 20 in the hall 2, the impulse responses can be recorded and used in modified or Hall added form as a given impulse response function for the convolution. Other improvements are also possible by adding or removing certain reverb portions of these impulse response functions.
  • the sound system according to the invention can also be easily adapted to spatial changes in the concert hall 2.
  • Corresponding predetermined impulse response functions for use in speech events, chamber music, orchestral music or lecture events can thus be kept flexible.
  • the sound system according to the invention can also respond to spatial changes by drawing in dividing walls.
  • the Fig. 3 shows a preferred embodiment of the sound system 100 according to the invention.
  • a microphone 4 for receiving a sound signal E1, which is fed to a folding device 5.
  • the folding device 5 performs convolutions of impulse response functions stored in the coupled memory with the recorded recording signal E1.
  • the six corresponding convolution signals L1, L2, L3, L4, L5, L6 are fed to the six inputs 21-26 of a switching device 7.
  • the switching device 7 has four outputs 27, 28, 29, 30, on which four folding signals M1, M2, M3, M4 can be tapped off.
  • the switching device 7 switches, for example, in a stochastic change the folding signals L1-L6 to the outputs 27-30.
  • two of the optimized sound signals are each over a respective Umschaltzeitintervall .DELTA.t N1-N6 not switched through.
  • the assignment of the signals L1-L6 present at the inputs of the switching device to the signals M1, M2, M3, M4 which can be tapped off at the outputs 27, 28, 29, 30 of the switching device 7 is preferably repeated periodically with a period duration of approximately 200 ms to 500 ms changed a given algorithm.
  • two signals L1-L6 are masked out.
  • On adjacent outputs 27, 28, 29, 30 of the switching device 7 are therefore never on the one hand never signals that were generated by means of a convolution with the same impulse response function, and on the other hand, the assignment of the signals differs on the outputs 27, 28, 29, 30 in successive following switching time intervals.
  • Two of the signals L1-L6 are also hidden here, in each case other signals are hidden in successive Umschaltzeitintervallen.
  • a random selection of the signals not to be switched through or to be hidden may be mentioned for each switching time interval.
  • the corresponding convolution signals M1-M4 are coupled to inputs 31, 32, 33, 34 of a level adjustment device 35.
  • This dynamic level adjustment means 35 provides at its outputs 36-39 weighted convolution signals N1, N2, N3, N4.
  • weighting factors w1, w2, w3, w4 are shown as a function of time t.
  • the respective periodic level reduction by the weighting factors wi (t) is set in time such that a corresponding time constant for the reduction of the respective level is selected such that it essentially corresponds to a time constant for a feedback building up in the room.
  • the periodicity of the level decreases of ⁇ ⁇ ⁇ t is thus chosen differently compared to the Umschaltzeitintervalle. Only the orders of magnitude of the time constants for the level reduction and the switching time intervals are similar.
  • stochastic or random level reductions can also be imposed on the convolution signals M1, M2, M3, M4.
  • the dynamic level-lowered Convolution signals N1, N2, N3, N4 are finally fed to a delay device 40, in which the level-matched folding signals N1, N2, N3, N4 are each subjected to delays and as loudspeaker signals 01, 02, 03, 04, 05 to amplifier devices 41, 42, 43 , 44, 45 are guided.
  • amplifier devices 41, 42, 43 , 44, 45 To the amplifier device 41-45 finally arranged in the corresponding room or concert hall speakers 46-55 are connected.
  • the respective delay times ⁇ tij are adapted to the position of the respective loudspeakers 46-55, to which the respective loudspeaker signal is supplied. Furthermore, the individual level-matched convolution signals N1, N2, N3, N4 can also be provided with a weighting factor aij. As a rule, it suffices to achieve a superimposition of the level-matched convolution signals N1-N4 directly at the listener location by the reproduction by loudspeakers 46-55. In this case, the delay means only serves to delay the convolution signals N1-N4. An additional superposition according to equation (3), however, may be advantageous if loudspeaker signals reproduced at certain listener seats are perceivable, which are generated by different branches of an extended sound system. A corresponding development with three branches is described below in de FIG. 5 explained in more detail.
  • the loudspeaker signals 01-05 are then respectively adapted so that at virtually every listener seat of the hall a direct sound signal which propagates through the transmission medium (air) in the hall from the sound source, always arrives before the additional sound signal generated by the sound signal.
  • the speakers are then preferably arranged so that a listener seat is operated by at least three speakers or recorded echo signals.
  • the described sound system which is operated by the method according to the invention, on the one hand has a very high level of feedback reliability due to the time and direction variance the recorded signals via the speakers.
  • a listener's psychoacoustically attains an increase in attention, as does, for example, the change in direction of the sound as a result of a movement of a musician when playing the respective instrument.
  • the audience sense is improved. Due to the temporal and spatial variance of the additionally recorded sound signals, the perceived reflections generated in this way become more conspicuous. Therefore, only a relatively low level of the recorded signals is required.
  • substantially real impulse responses whose fine structures are "improved” and thus correspond to the sound image or the perceived acoustics of a real room with a particularly pleasant acoustics
  • an overall sound field arises from the superposition of the natural surround sound and the added convolution signals and a feedback of this recording via the loudspeakers to the microphone inputs of the system, without disturbing feedbacks can occur.
  • the natural sound of a room with unsatisfactory perceived acoustics can be added by the sound system according to the invention additional natural sounding Hall to create a favorable sound.
  • the perceived acoustics of the room can be flexibly adapted by different impulse response functions, for example for language events, orchestral music, chamber music or solo voices.
  • the development 101 has three branches each with a convolver 5, 105, 205, a switching device 7, 107, 207, a dynamic level adjustment device 35, 135, 235 and an equalization device 40, 140, 240.
  • Memory 6, 106, 206 are respectively coupled to the folding devices 5, 105, 205 in which Prescribed pulse-response functions are stored.
  • Loudspeaker groups 146, 147, 148 are connected to the equalization devices 40, 140, 240 via amplifiers 141, 142, 143.
  • the branches work by the method according to the invention, as it is to the Fig. 3 has been described.
  • the sound signals G (t), H (t), K (t) supplied to the convolvers 5, 105, 205 are provided by a mixer 102 to the delayed and optionally filtered sound signals F1 recorded by microphones 303, 304, 305, 306, F2, F3 are supplied.
  • the microphones 303, 304, 305, 306 are arranged in the vicinity of a stage 3, on which one or more sound sources are present.
  • a stage 3 on which one or more sound sources are present.
  • these microphones 303, 304, 305 can be designed as high-quality studio condenser microphones with a directional characteristic "wide kidney". Thus, usually all sound sources in the stage area can be detected well.
  • the thus-detected sound signals E1, E2, E3, E4 are pre-amplified in an amplifier 307 and then supplied to the mixer device 102 as amplified sound signals F1, F2, F3, F4.
  • the mixer 102 delays and mixes the respective pre-amplified sound signals F1, F2, F3, F4.
  • the signal processing is similar, as already shown for example in equation (3), where the Oi (t) corresponds to the signals G (t), H (t) and K (t) and Nj (t) the pre-amplified sound signals Fj (t) corresponds.
  • the sound signals G (t), H (t), K (t) supplied to the convolution devices 5, 105, 205 thus have portions of all the signals E1, E2, E3, E4 received by the four microphones 303, 304, 305, 306, wherein signal components from more distant microphones 303, 304, 305, 306 are taken into account as a function of the sound propagation time between the microphone and the loudspeakers 146, 147, 148 coupled to the respective branch.
  • a loudspeaker group 146 of the first branch 100 is provided in a rear right area of the room, in the mixer and Retarder 102 signals of the microphones 205, 206 more delayed than the signals of the microphones 203, 204, which are arranged in the vicinity of the right stage area. Accordingly, the recorded sound signals of the microphones 303, 304, 305, 306 for loudspeakers 147 arranged in the middle of the room are then delayed approximately equally.
  • the mixer device 102 can be implemented, for example, as a digital signal processor, to which the pre-amplified signals F1, F2, F3, F4 are supplied in digitized form (for example via a high-resolution analog-to-digital converter, which is not shown here).
  • the further processing of the branch signals G (t), H (t), K (t) can then take place, for example, by means of a customary personal computer, to which the signals are coupled via a digital input of a sound card.
  • Today's computer technology makes it possible, for example, with several networked computers, practically parallel to fold the three times six different impulse response functions with the signals G (t), H (t), K (t) in real time and output again to a DSP in periodic change ,
  • the tasks of the folding device 5, 105, 205 and the switching device 7, 107, 207 can therefore be taken over by a particularly powerful PC or several networked PCs.
  • the corresponding post-coupled digital signal processor performs the tasks of dynamic level adjustment in the level adjusters 35, 135, 235 and the tasks of the delay and equalizer devices 40, 140, 240.
  • the latter further performs signal quality enhancing filtering or equalization to perform the transfer function to compensate for the speaker.
  • the corresponding output signals 01-05, T1-T5 and X1-X5 are digitally converted to analog (for example, with a conventional DA converter, which are not shown here) and the amplifiers 141, 142, 143 supplied.
  • the amplifiers each feed two to three loudspeakers at their outputs, which are regularly fitted distributed in the ceiling and wall area of the hall. In some cases, speakers are also placed below podiums or grandstands. The density of the speaker network should be chosen so that each listener seat is powered by 3-4 speakers.
  • the present invention thus provides a method and a sound system for improving a perceived acoustics of a room, which enables a natural and lively sound image based on real determined spatial properties by means of electro-acoustic interference.
  • a particularly pleasant surround sound is achieved.
  • the Einkopplungsraum the rehearsed additional optimized sound signals by switching in the switching devices unwanted localization of the additional signals is avoided by a listener and achieved a particularly high level of feedback security.
  • the room acoustics of event rooms can be significantly improved.
  • the present invention has been described in terms of preferred embodiments, it is not limited thereto, but modifiable in a variety of ways.
  • the number of parallel convolution signals can be changed as desired.
  • the use of the invention is not limited to closed rooms, but a corresponding sound system is also suitable for improving the room acoustics perception of open or semi-open venues.
  • synthesized impulse response functions can also be used as predetermined impulse response functions.
  • the waveforms shown in the figures are to be understood as exemplary only for explanation. Size specifications, for example, the volume of the rooms to be sounded or microphone distances, too, are also to be understood only as an example and virtually any change or scalable.

Claims (29)

  1. Procédé d'amélioration d'une acoustique perçue dans un local (1) comprenant les étapes suivantes :
    a) enregistrement d'au moins un signal acoustique (E1(t)) par l'intermédiaire d'un moyen d'enregistrement (4) situé à proximité d'au moins une source acoustique (3) dans le local (1) ;
    b) réalisation d'une convolution du signal acoustique enregistré (E1(t)) avec une fonction de réponse impulsionnelle (gi(t)) en vue de créer différents signaux acoustiques optimisés (L1(t), L2(t), L3(t)), la fonction de réponse impulsionnelle respective étant sélectionnée parmi une gamme de fonctions de réponse impulsionnelle prédéterminées pour le local (1) ;
    c) reproduction des signaux acoustiques optimisés (N1, N2, N3) à l'aide de moyens de reproduction sonore (14, 15, 16) disposés dans le local (1), un signal acoustique optimisé différent (N1, N2, N3) étant acheminé à chaque moyen de reproduction sonore (14, 15, 16), et les signaux acoustiques optimisés (N1, N2, N3) étant acheminés aux moyens de reproduction (14, 15, 16) à chaque fois après un intervalle de temps de commutation (Δt) prédéterminé de telle sorte que le même signal acoustique optimisé (N1, N2, N3) n'est acheminé à aucun des moyens de reproduction (14, 15, 16) à des intervalles de temps de commutation (Δt) successifs.
  2. Procédé selon la revendication 1, caractérisé en ce qu'au moins trois signaux sonores optimisés différents (L1, L2, L3) sont générés et au moins trois moyens de reproduction (14, 15, 16) sont prévus.
  3. Procédé selon la revendication 1 ou 2, caractérisé en ce qu'à l'étape c), on choisit un intervalle de temps de commutation (Δt) tellement réduit, et de préférence dans une plage d'environ 100-500 ms, que toute rétroaction changeant la tonalité est exclue entre un moyen de reproduction (14, 15, 16) et le moyen d'enregistrement (4).
  4. Procédé selon l'une des revendications précédentes, caractérisé en ce qu'à l'étape c), dans le cas d'un changement d'un premier à un second signal acoustique optimisé (N1, N2, N3) acheminé à un moyen de reproduction (14, 15, 16), le premier signal sonore optimisé (N1, N2, N3) est enchaîné en fondu dans le second.
  5. Procédé selon l'une des revendications précédentes, caractérisé en ce que les signaux acoustiques optimisés (N1, N2, N3) sont acheminés aux moyens de reproduction (14, 15, 16) dans une succession périodique ou une répartition stochastique.
  6. Procédé selon l'une des revendications précédentes, caractérisé en ce que l'intervalle de temps de commutation (Δt) est modifié dans le temps.
  7. Procédé selon la revendication 1, caractérisé en ce qu'à l'étape b), un nombre de signaux acoustiques optimisés (L1-L6) est généré, qui est supérieur au nombre des moyens de reproduction (14, 15, 16), et en ce qu'à l'étape c), au moins un des signaux acoustiques optimisés générés (L1-L6) n'est pas acheminé à un moyen de reproduction (14, 15, 16) pendant un intervalle de temps de commutation (Δt).
  8. Procédé selon l'une des revendications précédentes, caractérisé en ce que les fonctions de réponse impulsionnelle prédéterminées (gi(t)) sont sans corrélation entre elles.
  9. Procédé selon l'une des revendications précédentes, caractérisé en ce que le procédé de détermination d'au moins une des fonctions de réponse impulsionnelle prédéterminées (gi(t)) présente l'étape :
    i) mesure d'une fonction de réponse impulsionnelle (gi(t)) à un emplacement (17) dans le local (2).
  10. Procédé selon la revendication 9, caractérisé en ce que la fonction de réponse impulsionnelle mesurée (gi(t)) est enregistrée.
  11. Procédé selon la revendication 9 ou 10, caractérisé en ce qu'une étape supplémentaire est prévue :
    ii) suppression ou ajout de facteurs de réverbération (A3) à la fonction de réponse impulsionnelle mesurée (gi(t)) pour générer une fonction de réponse impulsionnelle prédéterminée correspondante.
  12. Procédé selon l'une des revendications 9 à 11, caractérisé en ce que les fonctions de réponse impulsionnelle prédéterminées (gi(t)) sont générées par la mesure de fonctions de réponse impulsionnelle à plusieurs emplacements (17 à 20) dans le local (2).
  13. Procédé selon l'une des revendications précédentes, caractérisé en ce que les fonctions de réponse impulsionnelle prédéterminées (gi(t)) sont générées de telle sorte que les premières composantes de réflexion correspondent sensiblement à des enregistrements microphoniques présentant une composante de son direct (DS) élevée.
  14. Procédé selon l'une des revendications précédentes, caractérisé en ce que les fonctions de réponse impulsionnelle prédéterminées (gi(t)) sont générées de telle sorte que les réverbérations correspondent sensiblement à des enregistrements microphoniques présentant un son diffus (DF) élevé.
  15. Procédé selon l'une des revendications précédentes, caractérisé en ce que les signaux acoustiques optimisés (M1 à M4) sont soumis à un changement de niveau périodique lors de la reproduction.
  16. Procédé selon la revendication 15, caractérisé en ce qu'une constante de temps du changement de niveau périodique est sélectionnée en fonction d'une constante de temps pour une structure de rétroaction dans le local.
  17. Procédé selon l'une des revendications précédentes, caractérisé en ce que les signaux acoustiques optimisés (N1, N2, N3, N4) sont retardés de telle sorte que les signaux acoustiques optimisés (O1 à 05) et les signaux acoustiques de la source acoustique (3) arrivent sensiblement en même temps à au moins un emplacement du local (2).
  18. Procédé selon l'une des revendications précédentes, caractérisé en ce qu'en cas de sources acoustiques multiples, les signaux acoustiques optimisés (N1 à N4) sont retardés de telle sorte qu'un son direct des sources acoustiques arrive à chaque emplacement du local avant une composante de réflexion d'un signal acoustique optimisé reproduit (O1 à 05).
  19. Procédé selon l'une des revendications précédentes, caractérisé en ce qu'à l'étape a), plusieurs signaux acoustiques individuels (E1 à E4) de la source acoustique (3) sont enregistrés sur plusieurs moyens d'enregistrement (303, 304, 305, 306) et les signaux acoustiques individuels correspondants (E1 à E4) sont mélangés les uns aux autres de façon à former un signal acoustique (G(t), H(t), K(t)) soumis à un traitement ultérieur, les signaux acoustiques individuels (E1 à E4) étant retardés en fonction des positions des moyens d'enregistrement (303, 304, 305, 306).
  20. Procédé selon l'une des revendications précédentes, caractérisé en ce les signaux acoustiques individuels (E1 à E4) ou les signaux acoustiques enregistrés sont filtrés pour réduire les composantes de fréquence supérieure.
  21. Système acoustique (1) pour un local (2), en particulier pour la réalisation du procédé selon au moins une des revendications 1 à 20, présentant :
    a) au moins un microphone (3) pour enregistrer au moins un signal acoustique (EI(t)) à proximité d'une source acoustique (3) ;
    b) un dispositif de convolution (5) permettant de convoluer le signal acoustique enregistré (El(t)) avec des fonctions de réponse impulsionnelle prédéterminées (gi(t)) pour générer au moins trois signaux acoustiques optimisés (L1, L2, L3) ;
    c) un dispositif de commutation (7) couplé au dispositif de convolution (5) ; et
    d) des haut-parleurs (14, 15, 16) couplés au dispositif de commutation (7) pour reproduire les signaux acoustiques optimisés (L1, L2, L3) ;
    e) le dispositif de commutation (7) acheminant toujours les signaux acoustiques optimisés (L1, L2, L3) aux haut-parleurs (14, 15, 16) après un intervalle de temps de commutation (Δt) de façon à ce que le même signal acoustique optimisé (L1, L2, L3) ne soit acheminé à aucun des haut-parleurs (14, 15, 16) dans des intervalles de temps de commutation (Δt) se succédant directement.
  22. Système acoustique (1) selon la revendication 21, caractérisé en ce qu'au moins trois signaux optimisés (L1, L2, L3) sont générés par une convolution du signal acoustique enregistré (E1) avec trois fonctions de réponse impulsionnelle différentes (gi(t)) et au moins trois haut-parleurs (14, 15, 16) sont prévus.
  23. Système acoustique (1) selon la revendication 21 ou 22, caractérisé en ce que les fonctions de réponse impulsionnelle prédéterminées (gi(t)) sont enregistrées dans une mémoire (6) couplée au dispositif de convolution (5).
  24. Système acoustique (1) selon l'une des revendications 21 à 23, caractérisé en ce qu'un dispositif d'adaptation de niveau (35) permettant la modification périodique des niveaux des signaux acoustiques optimisés (M1, M2, M3, M4) est monté en aval du dispositif de commutation (7).
  25. Système acoustique (1) selon l'une des revendications 21 à 24, caractérisé en ce qu'un dispositif de retardement (40) permettant de retarder les signaux acoustiques optimisés (M1, M2, M3, M4) est prévu.
  26. Système acoustique (1) selon l'une des revendications 21 à 25, caractérisé en ce que les haut-parleurs sont disposés de telle sorte dans le local (2) qu'au moins trois haut-parleurs (14, 15, 16) sont audibles par un auditeur à un emplacement (17) du local.
  27. Système acoustique (1, 100, 101) selon l'une des revendications 21 à 26, caractérisé en ce que sont également prévus :
    - plusieurs microphones (303, 304, 305, 306), qui sont disposés à proximité d'une estrade (3) du local (2) pour l'enregistrement de plusieurs signaux acoustiques individuels (E1 à E4) ; et
    - un dispositif de mélange (307, 102) permettant de mélanger les signaux acoustiques individuels (E1-E4) de façon à former le signal acoustique enregistré (G(t), H(t), K(t)).
  28. Système acoustique (101) selon l'une des revendications 21 à 27, caractérisé en ce que le système acoustique (1) présente plusieurs branches de signal avec au moins un signal acoustique enregistré (G(t), H(t), K(t)), un dispositif de convolution (5, 105, 205), un dispositif de commutation (7, 107, 207) et un groupe de haut-parleurs (146, 147, 148), les fonctions de réponse impulsionnelle prédéterminées correspondantes (gi(t)) étant choisies en fonction des positions des haut-parleurs.
  29. Salle polyvalente (2) présentant un système acoustique (1) selon l'une des revendications 21 à 28.
EP06777451A 2005-07-01 2006-06-23 Procede electro-acoustique Active EP1900250B1 (fr)

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PCT/EP2006/063515 WO2007003519A2 (fr) 2005-07-01 2006-06-23 Procede electro-acoustique

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NL8800745A (nl) * 1988-03-24 1989-10-16 Augustinus Johannes Berkhout Werkwijze en inrichting voor het creeren van een variabele akoestiek in een ruimte.
JP2569872B2 (ja) * 1990-03-02 1997-01-08 ヤマハ株式会社 音場制御装置
US5109419A (en) * 1990-05-18 1992-04-28 Lexicon, Inc. Electroacoustic system
DE4328620C1 (de) * 1993-08-26 1995-01-19 Akg Akustische Kino Geraete Verfahren zur Simulation eines Raum- und/oder Klangeindrucks
US5604839A (en) * 1994-07-29 1997-02-18 Microsoft Corporation Method and system for improving speech recognition through front-end normalization of feature vectors
JP3240947B2 (ja) * 1997-01-28 2001-12-25 ヤマハ株式会社 ハウリング検出器及びハウリングキャンセル装置
US6978027B1 (en) * 2000-04-11 2005-12-20 Creative Technology Ltd. Reverberation processor for interactive audio applications

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WO2007003519A2 (fr) 2007-01-11
DE502006006987D1 (de) 2010-07-01
EP1900250A2 (fr) 2008-03-19
WO2007003519A3 (fr) 2007-05-10
DE102005030855A1 (de) 2007-01-11

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