EP1900250A2

EP1900250A2 - Electroacoustic method

Info

Publication number: EP1900250A2
Application number: EP06777451A
Authority: EP
Inventors: Gunter Engel; Karlheinz Müller
Original assignee: Mueller Bbm GmbH
Current assignee: Mueller Bbm GmbH
Priority date: 2005-07-01
Filing date: 2006-06-23
Publication date: 2008-03-19
Anticipated expiration: 2026-06-23
Also published as: ATE468710T1; WO2007003519A2; DE502006006987D1; WO2007003519A3; EP1900250B1; DE102005030855A1

Abstract

Disclosed is a method for improving the perceived acoustics of a room. Said method comprises the following steps: at least one sound signal is recorded in the vicinity of at least one sound source located in the room via a recording means; the recorded sound signal is replicated once with one respective pulse response function to generate different optimized sound signals, the respective pulse response function being chosen among a selection of predefined pulse response functions for the room; the optimized sound signals are reproduced via sound reproducing means located in the room. A different optimized sound signal is fed to each sound reproducing means while the optimized sound signals are fed to the reproducing means following one respective switching interval such that the same optimized sound signal is fed to no reproducing means in directly succeeding switching intervals.

Description

Electro-acoustic method

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.

In event rooms, the room acoustic conditions usually only meet very limited requirements of different types of events. For example, 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. In musical events, on the other hand, a greater reverberation of the room is desired depending on the volume of the room, the style of the music and the orchestra's occupation between 1.0 and 2.5 s for classical music and even 2 to 10 s for sacred music. Furthermore, if necessary, lateral reflections to increase the spatiality and the perceived sound source size should be aimed at.

As a rule, 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. Within certain limits, 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.

Therefore, 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. At the same time, however, systems for electronic room acoustics control require a homogeneous approach to sound sources throughout the entire stage area and of high quality. 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.

In addition, the stage situation is very variable at opera performances or in multipurpose halls, whereby the natural room acoustic conditions can also change during the performance. A repositioning of recording microphones is then hardly possible.

Recently, therefore, electroacoustic systems for improving room acoustics have been proposed which deal with the feedback problem in various ways. From US Pat. No. 5,199,419, for example, an electroacoustic system is known in which sound signals recorded via a microphone are fed to a large number of Hall generators, each of which impress a synthetic reverb on the corresponding signals. These signals 20 are reproduced via a plurality of speakers, which are arranged in a correspondingly equipped, hall. The different Hall generators generate a random delay time and each have non-correlated transfer functions. The speakers are controlled so that adjacent speakers are not addressed by the same signal. Because, according to US Pat. No. 5,199,419, the amplitude

25 and time delays of the individual signals and the transfer functions used for the Hall generators are changed continuously in time and at random, so a feedback can be reduced. However, such synthetically generated additional reflections or reverberation may seem unnatural to a listener. The randomly synthesized reverberation generation also has the disadvantage that the special characteristics of the space, in

30 in which such an electroacoustic system is used by the randomization and artificial generation of the Hall are considered insufficient. It is therefore an object of the present invention to provide a method and system for improving a perceived acoustics of a room in which natural acoustical signals of the room are complemented time-synchronously with electroacoustically generated signals at every listener's seat, without a sound discoloring discernible sound. coupling arises or the additionally generated electro-acoustic signals can be localized by a listener.

This object is achieved by a method for improving a perceived acoustics of a room with the method steps of claim 1. Furthermore, an inventive sound system with the features of claim 17 solves this problem.

Accordingly, a method is provided with the following method steps:

a) recording at least one sound signal via a recording medium in the vicinity of at least one sound source in the room;

b) performing a convolution of the recorded sound signal, each having an impulse response function for generating different optimized sound signals, the respective impulse response function being selected from a selection of predetermined impulse response functions for the room;

c) reproducing the optimized sound signals via sound reproducing means arranged in the space, wherein a different optimized sound signal is supplied to each sound reproducing means, and wherein after each switching time interval the optimized sound signals are supplied to the reproducing means so that none of the reproducing means receive the same optimized sound signal in directly successive switching time intervals is supplied.

Accordingly, the sound system according to the invention provides for a room, in particular for carrying out this method according to the invention: a) at least one microphone for recording at least one sound signal in the vicinity of a sound source;

b) convolution means for convolving the recorded sound signal with predetermined impulse response functions to produce optimized sound signals;

c) a switching device coupled to the folding device; and

d) speakers coupled to the switching means for reproducing the optimized sound signals;

e) wherein the switching device in each case after a Umschaltzeitintervall the optimized sound signals to the speakers such that none of the speakers in directly successive Umschaltzeitintervallen the same optimized sound signal is supplied.

According to the invention, 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. According to the invention, the feedback reliability is achieved by subjecting the corresponding emission locations to a periodic change in switching time intervals. By switching the reproduced optimized

Sound signals, it is then not possible for a listener to locate the recorded sound. Rather, the inventive method leads to a particularly pleasant spatial sound. Furthermore, 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. By means of the invention, however, the recording of additional reverberation in the vicinity of the stage is also made possible.

Preferably, at least three different optimized sound signals are generated, and at least three reproduction means are provided. Preferably, the switching time interval is chosen so short that no sound-discoloring feedback is produced between a respective reproducing means and the recording means. As sound discoloring, unwanted level increases of limited frequency ranges are usually perceived. Already in a preliminary stage of the actual feedback, which can lead to a clearly perceptible whistling, for example, such increases in the signal level are disturbing even by local frequencies. According to the invention, such possible color discolorations are counteracted by the switching.

N 10 The switching time interval depends on the distance between the corresponding recording medium or recording microphone and the speakers 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.

In one embodiment of the method according to the invention, 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.

20

In a preferred embodiment of the method according to the invention is in a

Change from a first to a second reproducing means supplied optimized sound signal soft blends the respective first in the respective second optimized sound signal. Such a smooth transition creates a particularly homogeneous sound

25 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.

In a further development of the method, 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. Preferably, the predetermined impulse response functions are uncorrelated with each other.

In a development of the method, the steps may be provided for determining at least one of the predetermined impulse response functions:

i) measuring an impulse response function at a location in the room; and

, 10 ii) removing or adding Hall components to the measured impulse response functions to generate a corresponding predetermined impulse response function.

iii) storing the measured impulse response function and / or generated given impulse response function.

Most preferably, the predetermined impulse response functions are generated for multiple locations in the room. By first measuring the natural impulse response functions for 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, according to the invention a particularly natural

25 rather sound impression achieved.

Furthermore, the predetermined impulse response functions can be generated such that early reflection components in a respective impulse response function essentially correspond to a high direct-sound impulse response obtained by a microphone recording

30 and strong early reflections. Thus, 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 present. be taken. Likewise, the predetermined impulse response functions are preferably generated such that reverberation portions substantially correspond to microphone recordings having a high diffuse sound. Thus, preferably impulse response functions appearing as natural as possible are used for the convolution, which consist of impulse response functions of the surface and have additional components for direct sound and diffuse sound. It is equally possible to remove disturbing components in a measured impulse response function. Therefore, according to the invention, 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 "almost 10 natural" hall of the room is processed.

In a further development of the method according to the invention, the optimized sound signals during reproduction are subjected to a periodic level change. This periodic level change further reduces the risk of feedback and improves room perception for a listener. Preferably, 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.

In yet a further development of the inventive method, the optimized 20 sound signals are delayed in such a manner that the reproduced optimized sound signals and the _j sound signals arrival of the sound source at the same time at least one location of the space is substantially and preferably arrive, the sound signals of the sound source before.

In addition, if there are multiple sound sources, the optimized sound signals are so

25 delays 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.

In another preferred embodiment of the method, a plurality of individual sound signals of the sound source are recorded via a plurality of recording means and the corresponding ones

30 individual sound signals mixed to a recorded sound signal for further processing. The individual sound signals are delayed depending on the positions of the recording means. By using several recording media or microphones in Nearby, a particularly homogeneous sound can be achieved, taking into account the delay differences between the microphone and the sound source.

It is also advantageous if the single-tone signals or the recorded sound signals are filtered to reduce higher-frequency components. By filtering, 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, for example based on a realistic determination, 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. By means of three loudspeakers an improvement of the sound properties is already possible by the electro-acoustic 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 which is equipped with a sound system according to the invention. Further advantageous embodiments and modifications of the invention are the subject of the dependent claims and the following description of the embodiments. The invention will be explained in more detail below on the basis of exemplary embodiments with reference to the figures. It shows:

Fig.l: A schematic representation of the sound system according to the invention in one

Concert Hall;

Fig. 2: an example of a given impulse response function;

3 shows a preferred embodiment of the sound system according to the invention;

4 shows a representation of weighting functions for level reduction; and

Fig. 5: a preferred embodiment of the sound system according to the invention.

In the figures, unless otherwise stated, the same or functionally identical elements have been given the same reference numerals.

Fig. 1 shows an inventive sound system 1 for a room 2, for example, a concert or event hall. In the hall 2, a stage area 3 is provided, which serves here schematically as a sound source. In the vicinity of the sound source 3, a sound signal El (t) is recorded via a microphone 4. The recorded sound signal El (t) is fed to a convolver 5, which may be embodied 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. In the example shown here, the folding device 5 preferably performs three convolutions on the recorded sound signal El (t), each having an impulse response function gi (t), and thus generates three optimized sound signals L1 (t), L2 (t), L3 (t): (1) Li (t) = J dτ Ei (t) gi (t-τ)

The three optimized sound signals Ll, L2, L3 are supplied to a switching device 7, which is coupled to the folding device 5. 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 Ll, L2, L3 in such a way to their outputs 11, 12 and 13 that, for example during a first Umschaltzeitintervall the length .DELTA.t the signal Ll as a loudspeaker signal Nl on) 10 the first output 1 1 can be tapped, the second optimized sound signal L2 at the second output 12 is tapped as a second speaker signal N2 and the third optimized sound signal L3 at the third output 13 can be tapped off as the third speaker signal N3. In a second or subsequent switching time interval Ll is then fed to the output 12, L2 to the output 13 and L3 to the output 11. In a third switching interval 15, an assignment Ll to output 13, L2 to output 11 and L3 to output 12 could then follow. The corresponding loudspeaker signals Nl, N2, N3 are supplied to loudspeakers 14, 15 and 16, which are arranged in the concert hall 2 and reproduce the corresponding signals.

A listener in the room 2, for example at the location 17, on the one hand perceives the direct sound) from the stage 3 and on the other hand reverberates, which by the inventive

System 1 are discharged through the speakers 14, 15, 16. The subjective sound impression for a listener arises essentially from the original acoustic properties of the room 2 and the given impulse responses with which the recorded

25 signal E 1 (t) is folded.

In general, 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 curve at a specific location of the room as a functional

30 of the time when a delta-shaped acoustic wave is emitted at another location in the room. An exemplary impulse response gi (t) is shown in FIG. At time 0, 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. For example, the solid curve Al may correspond to the impulse response function of the space 2 shown in FIG. 1 for a transmitted impulse on the stage 3 and the received sound pressure at the position 20.

The first early components correspond to a direct sound DS, which propagates without any detours via reflections on possible surfaces to the receiver or listener. As early reflections that significantly influence the perceived clarity and distinctness of the sound signals, 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. As a particularly pleasant 15 rooms are usually perceived having an impulse response function, which corresponds approximately to a "fir tree", and having an envelope, such as the curve A2. The predefined 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

20 of FIG. 2, 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. Likewise, interfering components in a measured impulse response function can be removed and the predetermined impulse response functions thus generated can be used as the basis for the convolution in the convolution device 5.

Generally, by folding a sound signal of a first space with the impulse response of a second space, the sound characteristics of the second space can be impressed on the sound signals. The folding takes place as described in equation (1). Becomes a

However, if such a folded signal is reproduced without further processing in the second room, in this case, for example, in room 2, then feedback can occur and a listening Furthermore, he would be able to locate the additionally recorded reverb via the loudspeakers, which would normally be annoying.

According to the invention, the folded or optimized signals L 1, L 2, L 3 are therefore exchanged by the switching device 7. As 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 reverbated form as a predetermined impulse response function for the convolution. Other improvements are also possible by adding or removing certain reverb portions of these impulse response functions.

Overall, according to the invention, a particularly pleasant impulse response function is used. By storing such predetermined 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.

)

Fig. 3 shows a preferred embodiment of the sound system according to the invention

100. It is a microphone 4 for receiving a sound signal El is provided, which is fed to a folding device 5. The folding device 5 performs convolutions in the

25 coupled memory impulse response functions with the recorded recording signal El by. 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, at which four folding signals

30 Mental, M2, M3, M4 are tapped. The switching device 7 switches, for example, in a stochastic change the folding signals L1-L6 to the outputs 27-30. In this case, two of the optimized sound signals are each over a respective Umschaltzeitintervall .DELTA.t N1-N6 not switched through. In this case, 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 performed periodically with a period duration of approximately 200 ms to 500 ms changed a given algorithm. In the exemplary embodiment considered here, for example, two signals L1-L6 (different signals at successive switching time intervals) are masked out. On adjacent outputs 27, 28, 29, 30 of the switching device 7, on the one hand, there are never any signals which have been generated by means of a convolution with the same impulse response function, and on the other hand the assignment of the signals to the outputs 27, 28, x 10 29, 30 differs in successive switching time intervals. Two of the signals L1-L6 are also hidden here, in each case other signals are hidden in successive Umschaltzeitintervallen. By way of example only, 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 Nl, N2, N3, N4. The level adjuster 35 periodically lowers the respective convolution signals M1-M4 to the level-matched convolution signals N1-N4. This can be done, for example, by a time-dependent weighting factor

20 wi (t) happened:

)

(2) Ni (t) = Mi (t) wi (t)

Weighting factors w1, w2, w3, w4 as a function of time are shown by way of example in FIG

25 t shown. In this case, 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. Preferably, the periodicity of the level decreases of Δ τ ≠ Δ t is thus different compared to the switching time intervals

30 elected. Only the orders of magnitude of the time constants for the level reduction and the

Switching time intervals are similar. Alternatively, stochastic or random level reductions can be imposed on the convolution signals M1, M2, M3, M4. The dynamic level lowered convolution signals Nl, N2, N3, N4 are finally fed to a delay device 40, in which the level-matched convolution signals Nl, N2, N3, N4 are respectively subjected to delays and as loudspeaker signals O1, O2, 03, 04, 05 to amplifying devices 41, 42, 43, 44, 45 are guided. Finally, the loudspeakers 46-55 arranged in the corresponding room or concert hall are connected to the amplifier device 41-45. The loudspeaker signals 01-05 are generated in the delay device 40, for example as superpositions of the level-matched convolution signals N1, N2, N3, N4:

(3) Oi = X Nj (t -Δtij) aij,

) 10 where i = 1, ... 5 and j = 1, ... 4 is running. The respective delay times Δtij 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 an overlay of the level-matched convolution signals N1 -N4 directly at the listener location through 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 However, the equation (3) may be advantageous if perceptible at certain places listener reproduced speaker signals of different branches of a _ER-) 20 far Erten sound system are generated. A corresponding development with three branches is explained in more detail below in Figure 5.

The loudspeaker signals O1-O5 are then respectively adapted so that at virtually every listener seat of the hall a direct sound signal, which is transmitted via the transmission medium

25 (air) in the hall from the sound source propagates, 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.

30 The described sound system, which is operated with the method according to the invention, on the one hand has a very high level of feedback safety through the time and direction variance of the recorded signals via the loudspeakers. On the other hand, it is not possible for a listener to locate the recorded signals directly, since the periodic or even stochastic change of the sound signals 01-05 supplied to the loudspeakers after a respective switchover time interval, which may take, for example, 100-500 ms, causes the 5 sound irradiation locally changed. As a result, the sound appears natural and lively. As a result of this time and direction variance, 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. Thus, even without a level increase, the acoustic signals additionally brought in by the sound system according to the invention improve the perception of the listener.

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.

By using 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, in the event room equipped with a system according to the invention an overall sound field results from the overlaying of the natural one Surround sound and the added convolution signals and a

20 Reverse effect of this recording via the loudspeakers on the microphone inputs of the system without disturbing feedbacks. Thus, 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 understood

25 different preset impulse response functions versatile adapt, for example

Language events, orchestral music, chamber music or solo voices.

FIGS. 5A and 5B show a development of the sound system according to the invention. Essentially, the development 101 has three branches with one fa

30 processing device 5, 105, 205, a switching device 7, 107, 207, a dynamic

Level adjustment means 35, 135, 235 and an equalization means 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. Essentially, the branches operate according to the method of the invention as described for FIG.

The sound signals G (t), H (t), K (t) supplied to the convolution devices 5, 105, 205 are provided by a mixer device 102, to which the delayed and possibly filtered sound signals Fl, 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. When using the sound system according to the invention in an example 500Om ³ large multi-purpose auditorium, which is to be used for conventional acoustic amplified events as well as for theater or orchestral concerts, an arrangement of the microphones 303, 304, 305, 306 at a distance of about 2m before Stage and a height of 5m above the stage level. By way of example, these microphones 303, 304, 305 can be designed as high-quality studio condenser microphones with a directional characteristic "wide kidney." Thus, it is usually possible to detect all sound sources in the stage area well.

The sound signals E1, E2, E3, E4 thus detected are preamplified 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 Fl, 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 El, E2, E3, E4 received by the four microphones 303, 304, 305, 306, wherein signal components of more distant microphones 303, 304, 305, 306 as a function of the sound transit time between see microphone and the speakers coupled to the respective branch 146, 147, 148 are taken into account. For example, if 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. 5

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 be done, for example, N 10 by means of a conventional personal computer, 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 15, 105, 205 and the switching device 7, 107, 207 can therefore be carried out by a particularly powerful PC or several networked PCs.

The corresponding post-coupled digital signal processor (DSP) performs the tasks of dynamic level adjustment in the level adjustment means 35, 135, 235 and

20 of the delay and equalization devices 40, 140, 240. The latter further carry out the signal quality enhancing filtering or equalization to compensate for the transfer function of the loudspeakers. The corresponding output signals O1-O5, Tl -T5 and Xl -X5 are digitally converted to analog (for example, with a conventional DA converter, which are not shown here) and the amplifiers 141, 142, 143, respectively.

25

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 loudspeaker network should be selected in such a way that each listener's seat has 3-4 loudspeakers

30 is supplied. 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 on the basis of real-determined spatial properties by means of electro-acoustic influence. The convolution with predefined impulse response functions, which are constructed from measured impulse response functions of rooms which are as ideal as possible, achieves a particularly pleasant surround sound. By changing the Einkopp- direction of the rehearsed additional optimized sound signals by switching in the switching devices unwanted localization of the additional signals is avoided by a listener and a particularly high feedback security 110 reaches. By the method and system according to the invention, the room acoustics of event rooms can be significantly improved.

Although the present invention has been described in terms of preferred embodiments, it is not limited thereto, but modifiable in a variety of ways.

15 The number of parallel convolution signals can be changed as required. The use of the invention is not limited open spaces, but a corresponding sound system is also suitable for improving the room acoustics perception of open or semi-open venues. Furthermore, synthesized impulse response functions can also be used as predetermined impulse response functions. In the figures

20 ren shown waveforms are merely exemplary for explanation. _GroE) ßenangaben, for example, the volume of the spaces to be sonicated or microphone distances to are also exemplary only to understand and virtually any changeability or scalable.

LIST OF REFERENCE NUMBERS

1 sound system

5 2 event hall

3 stage

4 microphone

5 folding device

6 memory

> 10 7 switching device

8,9,10 entrance

11, 12, 13 output

14, 15, 16 speakers

17-20 listener position

15 21-26 entrance

28-30 output

31-34 entrance

35-39 output

40 Equalization device

20 41-45 amplifiers

J 46-55 speakers

100 sound system

101 sound system

102 mixers

25 105 folding device

106 memory

107 switching device

135 level adjusting device

140 delay device

30 141, 142, 143 amplifiers

146, 147, 148 speaker group

205 folding device 206 memory

207 switching device

235 level adjusting device

240 delay device

5 303-306 microphone

307 amplifiers

Al impulse response function

A2 envelopes

A3 Sound content α I o DS Direct sound

DF diffuse sound

E1-E4 microphone signal

F1-F4 pre-amplified signal

G, H, K recorded sound signal

15 L1-L5. P1-P6 convolution signal

U1-U6 convolution signal

M1-M4, Q1-Q4 switched convolution signal

Vl -V4 switched convolution signal

N1-N4, R1-R4 level-adjusted signal

20 Y1-Y4 level-adjusted signal

01-05, Tl -T5 Loudspeaker signal

)

X1-X5 speaker signal

Δt switching time interval wl-w4 weighting function

Claims

claims

1. A method for improving a perceived acoustics of a room (1) with the method steps:

a) recording at least one sound signal (El (t)) via a recording means (4) in the vicinity of at least one sound source (3) in the space (1);

b) performing in each case a convolution of the recorded sound signal (El (t)), each having an impulse response function (gi (t)) for generating different optimized sound signals (Ll (t), L2 (t), L3 (t)), wherein the respective impulse response function is selected from a selection of predetermined impulse response functions for the room (1);

c) reproducing the optimized sound signals (N1, N2, N3) via sound reproduction means (14, 15, 16) arranged in the space (1), each sound reproduction means (14, 15, 16) having a different optimized sound signal (N1, N2, N3 ), and wherein in each case after a predetermined switching time interval (Δt) the optimized sound signals (Nl, N2, N3) are supplied to the display means (14, 14, 16) such that none of the display means (14, 15, 16) in directly successive Umschaltzeitintervallen (.DELTA.t) the same optimized sound signal (Nl, N2, N3) is supplied.

2. The method according to claim 1, characterized in that at least three different optimized sound signals (Ll, L2, L3) are generated and at least three playback means (14, 15, 16) are provided.

3. The method according to claim 1 or 2, characterized in that in step c) the switching time interval (Δ t) as short and preferably in a range of about 100-500ms is selected that between a display means (14, 15, 16) and the recording means (4) a sound discoloring feedback is excluded.

4. The method according to any one of the preceding claims, characterized in that in step c) in a change from a first to a second, a reproduction means (14, 15, 16) supplied optimized sound signal (Nl, N2, N3) each first soft in the respectively second optimized sound signal (Nl, N2, N3) is faded.

5. The method according to any one of the preceding claims, characterized in that the optimized sound signals (Nl, N2, N3) in a periodic order or randomly stochastically distributed to the reproducing means (14, 15, 16) are performed.

6. The method according to any one of the preceding claims, characterized in that the switching time interval (.DELTA.t) is changed over time.

7. The method according to claim 1, characterized in that in step b) a number of optimized sound signals (Ll-L6) is generated, which is greater than the number of the reproducing means (14, 15, 16), and that in step c) during a switching interval (.DELTA.t) at least one of the generated optimized sound signals (L1-L6) not to a playback means 14, 15, 16) is guided.

8. The method according to any one of the preceding claims, characterized in that the predetermined impulse response functions (gi (t)) are uncorrelated with each other.

9. The method according to any one of the preceding claims, characterized in that the method for determining at least one of the predetermined impulse response functions (gi (t)) comprises the step: i) measuring an impulse response function (gi (t)) at a location (17) in the space (2).

10. The method according to claim 9, characterized in that the measured impulse response function (gi (t)) is stored.

11. The method according to claim 9 or 10, characterized in that a further step is provided:

ii) removing or adding Hall components (A3) to the measured impulse response function (gi (t)) to produce a corresponding predetermined impulse response function.

12. The method according to any one of claims 9-11, characterized in that the predetermined impulse response functions (gi (t)) are generated by measuring impulse response functions at a plurality of locations (17-20) in the space (2).

13. The method according to any one of the preceding claims, characterized in that the predetermined impulse response functions (gi (t)) are generated such that early reflection components substantially correspond to microphone recordings with a high direct sound content (DS).

14. The method according to any one of the preceding claims, characterized in that the predetermined impulse response functions (gi (t)) are generated such that reverberant shares substantially microphone recordings with a high diffuse sound (DF) correspond.

15. The method according to any one of the preceding claims, characterized in that the optimized sound signals (M1-M4) are applied during playback with a periodic level change.

16. The method according to claim 15, characterized in that a time constant of the periodic level change is selected in dependence on a time constant for a feedback structure in the room.

17. The method according to any one of the preceding claims, characterized in that the optimized sound signals (Nl, N2, N3, N4) are delayed such that the reproduced optimized sound signals (O1-O5) and the sound signals of the sound source (3) at least one Place of the room (2) arrive substantially simultaneously.

18. The method according to any one of the preceding claims, characterized in that at several sound sources, the optimized sound signals (N1-N4) are delayed such that a direct sound of the sound sources at each location of the room in time before a reflection component of a reproduced optimized sound signal (O1 -O5) arrives.

19. The method according to any one of the preceding claims, characterized in that in step a) a plurality of individual sound signals (El -E4) of the sound source (3) via a plurality of recording means (303, 304, 305, 306) are recorded and the corresponding individual sound signals (E1- E4) are mixed together to form a sound signal (G (t), H (t), K (t)) for further processing, the individual sound signals (E1-E4) being dependent on the positions of the recording means (303, 304, 305, 306 ) be delayed.

20. The method according to any one of the preceding claims, characterized the single-tone signals (E1-E4) or the recorded sound signals are filtered to reduce higher-frequency components.

21 sound system (1) for a room (2), in particular for carrying out the method according to at least one of claims 1 - 20, with:

a) at least one microphone (3) for recording at least one sound signal (El (t)) in the vicinity of a sound source (3);

b) convolution means (5) for convolving the recorded sound signal (E 1 (t)) with predetermined impulse response functions (gi (t)) to produce at least three optimized sound signals (L 1, L 2, L 3);

c) a switching device (7) coupled to the folding device (5); and

d) loudspeakers (14, 15, 16) coupled to the switching device (7) for reproducing the optimized sound signals (L1, L2, L3);

e) wherein the switching device (7) in each case after a switching time interval (Δt) the optimized sound signals (Ll, L2, L3) the speakers (14, 15, 16) such that none of the speakers (14, 15, 16) in directly successive Umschaltzeitintervallen (.DELTA.t) the same optimized sound signal (Ll, L2, L3) is supplied.

22 sound system (1) according to claim 21, characterized in that at least three optimized sound signals (Ll, L2, L3) by a convolution of the recorded sound signal (El) with three different impulse response functions (gi (t)) is generated and at least three Loudspeakers (14, 15, 16) are provided.

23. Sound system (1) according to claim 21 or 22, characterized the predetermined impulse response functions (gi (t)) are stored in a memory (6) coupled to the convolution device (5).

24 sound system (1) according to any one of claims 21 - 23, characterized in that the switching means (7) is a level adjusting means (35) for periodically changing the level of the optimized sound signals (Ml, M2, M3, M4) connected downstream.

25. Sound system (1) according to any one of claims 21 - 24, characterized in that a delay device (40) for delaying the optimized sound signals (Ml, M2, M3, M4) is provided.

26 sound system (1) according to any one of claims 21-25, characterized in that the loudspeakers are arranged in the space (2), that in each case at least three speakers (14, 15, 16) in each case a place (17) in the room is perceived by a listener.

27. Sound system (1, 100, 101) according to any one of claims 21-26, characterized in that are further provided:

- a plurality of microphones (303, 304, 305, 306), which are arranged in the vicinity of a stage (3) of the space (2) for receiving a plurality of individual sound signals (El -E4); and

- Mixing means (307, 102) for mixing the single sound signals (E1-E4) to the recorded sound signal (G (t), H (t), K (t)).

28. Sound system (101) according to any one of claims 21 - 27, characterized in that the sound system (1) comprises a plurality of signal branches each having at least one recorded sound signal (G (t), H (t), K (t)), a convolution device (5, 105, 205), a switching device (7, 107, 207) and a group of loudspeakers (146, 147, 148), wherein the respective predetermined impulse response functions (gi (t)) are selected in dependence on the positions of the loudspeakers.

29 event hall (2) with a sound system (1) according to any one of claims 21 - 28.