Method for eliminating room modes, and digital signal processor and loudspeaker therefor

Abstract

The invention relates to a method for eliminating room modes Ñ. The method consists substantially in using a digital signal processor (60) to generate and store a filter W*(z) in a two-stage characteristic measurement, which filter characterises and maps the sound changes of a user signal N emitted into a room (10) by a main loudspeaker (21), including the room modes Ñ generated in this way. The filter W*(z) generates a changed signal Ñ- from the digitally available user signal N, which is played by the main loudspeaker (21), to eliminate the room modes Ñ, which changed signal is played by a correction loudspeaker. The two signals cancel each other out in the room (10). Because passage through the filter W(z) requires a certain time dt, the original signal N remains completely audible in the room (10) and cannot be eliminated. The signal Ñ- can therefore eliminate only the portion of the soundwaves in the room (10) that is still present after this time dt. This ensures that the original signal N is audible in the room (10) completely and unchanged in comparison with the source, while the long-lasting room modes Ñ are eliminated after a short time dt. The main and correction loudspeakers (21, 22) may be the same loudspeaker (23). The invention also relates to a digital signal processor (60) provided for this purpose and to a loudspeaker (21, 22, 23).

Description

Technical field

[0001] The invention relates to methods for eliminating room modes that form as resonances in a room when a user signal is played through a main loudspeaker. Technical background

[0002] In every room, for example in recording studios, resonances form at certain frequencies, known as room modes. These are distorted in level and more persistent compared to other sounds played at the same level, and are perceived as disturbing.

[0003] A room mode can develop when half a wavelength of a tone, or an integer multiple thereof, corresponds exactly to the length, width, or another important dimension of the room in which the tone is produced. A good analogy is the excitation of a guitar string, which produces fundamental tones and overtones. The frequencies of problematic room modes are in the bass range, with higher overtones being less pronounced.

[0004] The intention is that the original tones of a played user signal should be heard completely and correctly across all frequencies and then decay as if the room were infinitely large. No tones of the original signal should be suppressed, boosted, or altered in time. Sounds reflected from the walls of the room are generally permitted, except at the frequencies of the room modes.

[0005] Known solutions to this problem use a filter that adjusts the volume of problematic tones in the audio output according to the problem. However, these solutions alter the original sound of the audio signal, which is why they should be avoided.

[0006] In US 10490180 or WO 2017 / 037341 A1, a first digital signal describing a system impulse response caused by a loudspeaker is attenuated and added with a time delay to a second digital signal describing a canceling impulse response. A delay value must be determined for this process.

[0007] The disadvantage of this or any other method that determines a delay value is the limitation that the resulting correction response is designed for only a few specific room modes, which can be eliminated. However, every room contains a multitude of different room modes, for example, longitudinally and transversely, vertically, and diagonally. Different room modes are dominant depending on the frequency. Each room mode generates its own impulse response with a different, inherent delay time. This is determined by the type of room mode and the position of the loudspeaker and microphone. Naturally, a time-delayed correction signal can only cancel out a small subset of all modes.

[0008] In EP 2357847, a delay response to be equalized is assigned to a loudspeaker at a predefined position within a listening room. Filter coefficients are then calculated for all-pass filters, each positioned upstream of this loudspeaker, with the all-pass filter having a transfer characteristic such that the corresponding delay response matches a predefined target delay.

[0009] These solutions typically have the disadvantage that a precise time delay must be determined so that the Active Noise Cancelling / Control signal (ANC signal) is superimposed on the original signal at the correct phase. Furthermore, the computing power required is very high because the ANC signal must be calculated very quickly and provided with high resolution in order to output the correction signal in time. Description of the invention

[0010] Therefore, the object of the present invention is to describe a method that can play back the complete original tone sequence of a given signal in a room while preventing the formation of room modes. Preferably, this should be possible with relatively low computing power, so that timely correction of the original signal can be carried out, using commercially available processors such as those commonly used in consumer goods.

[0011] Further objectives of the invention are to describe a digital signal processor (DSP) with which this method can be carried out, and a loudspeaker that can fill the room with sound from the user signal or a correction signal without room modes disturbing the acoustics in the room.

[0012] The tasks are solved by the claims in the corresponding categories. Advantageous methods and embodiments are described in the respective dependent claims.

[0013] The inventive embodiments are described in connection with the figures.

[0014] The solution of the invention essentially consists in a two-stage characteristic measurement using a digital signal processor to generate and store a filter W*(z), which characterizes and maps the sound changes of a user signal N emitted into a room by a main loudspeaker, including the room modes Α thus generated. The filter W generates a modified signal Α- from the digitally available user signal N, which is played back by the main loudspeaker, to cancel out the room modes Α-. This modified signal is then played back by a correction loudspeaker. The two signals cancel each other out in the room. Since the passage through the filter W requires a certain time dt, the original signal N remains fully audible in the room and cannot be canceled out. Consequently, the signal Α- can only cancel out the portion of the sound waves in the room that is still present after this time dt.This ensures that the original signal N is fully audible in the room, while the long-lasting room modes Ñ are canceled out after a short time dt. The main and correction loudspeakers can be one and the same loudspeaker.

[0015] The user signal N must be available as a digital signal or be digitized from an analog signal in order for the inventive method to be carried out with it.

[0016] In the first stage of the characteristic measurement, the acoustic characteristics of a digital signal A are simulated and stored in an intermediate filter S'*(z) after it has been played back by the correction loudspeaker in this room. For this purpose, signal B is received and analyzed by a microphone in the room.

[0017] In the second stage, a signal C is played into the room from both the main and the correction loudspeaker and received by the microphone as signal D. A filter W(z) is then generated, which modifies signal C before it is sent to the correction loudspeaker. The filter W is then adjusted until the sound is virtually eliminated at the microphone. This is possible because the characteristic of the correction loudspeaker is already known as S'*(z). This filter is then stored as W*(z) and used for the application of the procedure. Brief explanation of the characters

[0018] The invention will be explained in more detail below in connection with the drawings. The drawings show: Fig. 1 A digital signal processor (DSP) for use in a first characteristic measurement when separate main loudspeakers and correction loudspeakers are used; Fig. 2 A digital signal processor with its connections for use in a second characteristic measurement when separate main loudspeakers and correction loudspeakers are used; Fig. 3 A digital signal processor with its connections for use in the method with a user signal when separate main loudspeakers and correction loudspeakers are used; Fig. 4 A digital signal processor with its connections for use in a first characteristic measurement when a common loudspeaker is used; Fig. 5 A digital signal processor with its connections for use in a second characteristic measurement when a common loudspeaker is used; Fig.6. A digital signal processor with its connections for use in the method with a user signal when a common loudspeaker is used; Fig. 7. A schematic diagram of a digital signal processor with its connections for use in the . Figures 1 to 3 , when separate main speakers and correction speakers are used; Fig. 8 a schematic diagram of a digital signal processor with its connections for use in the Figures 4 to 6 , when a common loudspeaker is used; Fig. 9 a room set up for carrying out the procedure; Fig. 10 a digital signal processor with attached audio electronics, shown schematically. Ways to implement the invention

[0019] In Fig. 9A room 10 is schematically depicted in which the inventive method can be carried out, for example, a recording studio. A concert hall or a living room can also serve as room 10. As a recording studio, the walls 13 bounding room 10 generally do not have right angles to each other in order to prevent room modes as much as possible. Room 10 can be equipped with optional additional loudspeakers 12, which are irrelevant and have no influence on the inventive method. Furthermore, furniture such as a mixing console and a person's seat 11, as well as computers and screens of any kind, can be arranged in this room 10. A typical size for such a room 10 is 10–100 m², although the method can also be used effectively in larger and smaller rooms 10.

[0020] For the method according to the invention, a main loudspeaker 21 and a correction loudspeaker 22 must be arranged in the room 10, wherein these can be configured as separate loudspeakers or, alternatively, as a single, combined loudspeaker 23. Furthermore, a microphone 30 must be arranged in the room, wherein the loudspeaker(s) 21, 22, 23 and the microphone 30 are connected to a digital signal processor 60 via appropriate audio electronics.

[0021] In Fig. 10The digital signal processor 60 with the connected audio electronics 40 is shown schematically. This includes, for example, a digital-to-analog converter 41 of any type for each of the loudspeakers 21, 22, 23 used, optionally a preamplifier 42 and a power amplifier 43, while the microphone 30 is connected to a microphone amplifier 44 and to an analog-to-digital converter 45, with lines 46 serving as the connections. The converters 41, 45 are finally connected to the designated terminals on the signal processor 60. Alternatively, the necessary audio electronics 40 can be integrated into the signal processor 60.

[0022] Only one loudspeaker 21, 22, 23 is listed in each case, but there can also be several loudspeakers 21, 22, 23 with the same functions, which are also connected to the required audio electronics 40.

[0023] With the in Fig. 9The device connected to the signal processor 60, as shown, allows the loudspeakers 21, 22, 23 to emit sound waves 24 into room 10, which can be detected by the microphone 30 and forwarded as a signal to the signal processor 60. In addition, room modes Ñ are formed in room 10, one of which is shown schematically here as a dashed sine wave. Fig. 9 Two alternative, preferred positions for the microphone 30 are shown, but only one microphone 30 is used, and only during the performance of characteristic value measurements. When using the method with a user signal N, no microphone 30 is required. For this, only the signal processor 60 and one or two loudspeakers 21, 22, 23 connected to it with the necessary audio electronics 40 are needed.

[0024] In principle, the method can be implemented using the two possibilities mentioned here: In the first variant, a main loudspeaker 21 and a separate correction loudspeaker 22 are provided; in the second, these are combined into a single loudspeaker 23. The first variant is described in the Figures 1-3 and 7, the second in the Figures 4-6 and 8 more precisely described.

[0025] The inventive method essentially comprises a first and a second measurement of characteristic values, as described in the Figures 1 and 2 resp. 4 and 6 for the first and second variants, as well as the actual use with a user signal, as shown in the Figures 3 and 6 .

[0026] The Figures 7 and 8 The two variants of the signal processor 60 are shown as a circuit diagram, for carrying out all process steps in each case.

[0027] The invention is described in detail below: The method according to the invention serves to eliminate room modes Ñ, which form as resonances in a room 10 when a digital user signal N is played back through a main loudspeaker 21. It is characterized by the following steps: a. Setting up and carrying out characteristic measurements, by i. positioning a main loudspeaker 21 and a correction loudspeaker 22 in a room 10, for example in a recording studio, wherein these loudspeakers 21, 22 may be two separate or a common loudspeaker 23; positioning a microphone 30 in this room 10; Provision of a digital signal processor 60 with a signal input 61 for inputting and processing digital signals A, C, N, a first 62 and a second 63 loudspeaker output for the main loudspeaker 21 and the correction loudspeaker 22, which can be combined into a common loudspeaker output 64 for the common loudspeaker 23, and a microphone input 65 for the microphone 30, connection of the signal processor 60, the loudspeaker(s) 21, 22, 23 and the microphone 30 to an audio electronics unit 40, for generating and capturing sound waves 24 by means of loudspeakers 21, 22,23 and microphone 30; ii. Performing a first characteristic measurement in which a first transfer function S(z), which maps the change of a digital signal A after it has been recorded at microphone 30 on a secondary path S, which is played back as sound waves 24 via the correction loudspeaker 22 or the common loudspeaker 23 and recorded as a digital signal B [AS(z)=B], is modeled by a variable electronic filter S'(z), using an LMS (Least Mean Square) module for performing a numerical gradient method, preferably by means of the filtered-x LMS method, which generates the variable filter S'(z) based on knowledge of the original signal A and adjusts it until the original signal A, after passing through this filter S'(z),the detected signal B at the end of the secondary path S corresponds to and cancels it out as far as possible at an electronic subtractor 75 [AS(z) - AS'(z) = 0]; iii. Storing this electronic filter S'*(z), which thus becomes unchangeable; iv. Performing a second characteristic measurement in which a second transfer function Pz, which represents the change of a digital signal C after it has been recorded at the microphone 30 on a primary path P, which is played back as sound waves 24 via the main loudspeaker 21 or the common loudspeaker 23 and recorded as a digital signal D [CP(z)=D], is partially replicated by a variable electronic filter W(z), using the LMS module to perform a numerical gradient method, preferably using the filtered-x LMS method, which, based on the knowledge of the original signal C after it has passed through the stored filter S'*(z),The variable filter W(z) is generated and adjusted until the original signal C, when passing through this variable filter W(z) and subsequently the secondary path S, corresponds to the negative of the originally acquired signal D from the primary path P and is accordingly minimized when combined at the microphone 30 [CP(z) - CWS(z) ≈ 0]; v. Storing the electronic filter W*(z), which thus becomes unchangeable; b. Setting up and using the method with a user signal N, by i. positioning the main loudspeaker 21 and the correction loudspeaker 22, or the common loudspeaker 23, at the same locations in the same room 10 as in the characteristic measurements, with the same audio electronics 40 required for this purpose and connecting to the digital signal processor 60 as in step a.; ii. Routing the digital user signal N to the first or common output 62, 64 and playback through the main or common loudspeaker 21, 23,where room modes Ñ form in room 10; iii. simultaneous routing of this user signal N through the last stored filter W*(z) in the signal processor 60 and subsequent routing to the second or common output 63, 64, and, due to the time dt required by the filter W*(z), delayed playback of this filtered user signal (Ñ-) by the correction or common loudspeaker 22, 23; iv. thereby eliminating the room modes Ñ of the digital user signal N that are still present in room 10 after the time delay dt.

[0028] To explain, in step a., room 10 is first set up. It is important to ensure that loudspeakers 21, 22, and 23 always remain in the same location. The characteristic measurements only need to be performed once. Afterwards, any user signals N can be played back until the room geometry changes in relation to the loudspeaker positions. Therefore, if, for example, the room is divided or the loudspeaker position is changed, new characteristic measurements will be necessary. For this reason, the position of loudspeakers 21, 22, and 23 should be chosen carefully.

[0029] Since room modes Ñ are usually low-frequency, it is advisable to use subwoofers as loudspeakers 21, 22, 23, as these are suitable for playing low-frequency tones.

[0030] The microphone 30 can, in principle, be positioned anywhere in room 30. However, it has proven advantageous to position the microphone 30 in step a. at a location where either a person 11 is expected to be in step b., or near a wall 13 of room 10 that is far away from the main loudspeaker 21.

[0031] The goal of the first measurement of key performance indicators, as shown in Fig. 1 and 4 , is the determination and storage of the unchangeable electronic filter S'*(z), which is used for the second characteristic measurement, represented in Fig. 2 and 5This is required. For this purpose, the variable filter S'(z) is modified and adjusted by the LMS module until the signal output from subtractor 75 [AS'(z) - B] at the second input 72 into the LMS module is minimal, ideally zero. As long as this is not the case, the LMS module continues to modify the variable filter S'(z). The original signal A, which enters the LMS module at the first input 71, serves as its reference. The variable filter S'(z) electronically maps the transfer function S(z) on the secondary path. This means that a signal A passing through the filter S'(z) changes in the same way as the signal A passing through the secondary path via loudspeaker 22, 23 and microphone 30, temporarily as a sound wave 24. This change also includes all spatial distortions and room modes Ñ, which propagate in space 10 and continue to resonate long after an impulse has been emitted.

[0032] The simulation is set using the LMS module and takes approximately 10 to 30 seconds. The determined, adjustable filter S'(z) can then be saved as a fixed filter S'*(z). It is only needed for the second characteristic measurement, not for the subsequent operational phase. Any signal can be used as signal A, provided it has a sufficient frequency component across all relevant low frequencies. White or, preferably, pink noise has proven suitable.

[0033] The signal processor 60 can then be configured for this second characteristic measurement. The variable filter W(z) now occupies the space previously occupied by the variable filter S'(z). Its output leads to the second or common output 63, 64 and finally to the correction loudspeaker 22 or the common loudspeaker 23, as shown in Fig. 2 and 5shown. Alternatively, another signal processor 60 can be used, which is set up for this second characteristic measurement.

[0034] The goal of this second measurement of key performance indicators, as shown in Fig. 2 and 5 , is the determination and storage of the unchangeable electronic filter W*(z), which is ultimately used according to Fig. 3 and 6 This second characteristic measurement is required. It is completed when the signal D, which is sent from microphone 30 to the LMS module, is as effectively eliminated as possible.

[0035] It should be noted that the signal D transmitted by microphone 30, which originally originates from the sound waves 24 from the primary path P, CP(z), is soon superimposed and therefore altered by the sound waves 24 from the secondary path S through the filter W(z). The best possible cancellation of signal D is achieved when the sound waves generated by the correction loudspeaker 22 compensate as much as possible for those generated by the main loudspeaker 21. This, in turn, means that the signal C on the primary path P is altered as similarly as possible to how it passes through the filter W(z) and then along the secondary path S. Since this secondary path S is already known from the first characteristic measurement in the form of S'*(z), the resulting filter W*(z) can be determined in the same way as S'*(z) was previously determined.White or, preferably, pink noise can again be used as the output signal C; the procedure takes about as long as the first measurement of the characteristic value.

[0036] The LMS module can only recognize and react to causal relationships in the signals from its two inputs 71 and 72. It modifies the electronic filter W(z) in such a way that no components remain in signal D that have a causal relationship with the signal from filter position 74. Therefore, for example, ambient noise, which only enters input 72 into the LMS module, is irrelevant and has no influence on the characteristic measurement or on the modification of the electronic filter W(z).

[0037] The passage of signal C through the electronic filter W(z) requires a certain time dt. Therefore, the signal in the secondary path S is emitted by the correction loudspeaker 22, 23 later than in the primary path P by the main loudspeaker 21, 23. Thus, the signal in the secondary path S cannot eliminate the original signal C in the primary path P because it arrives too late. The room modes Ñ, on the other hand, continue to resonate in room 10 for a long time and propagate there. They have a causal relationship with the original signal that caused them and can therefore be eliminated by the correction loudspeaker 22, 23.

[0038] The signal C takes a time dt to pass through the filter W(z) and also a time dt s to traverse the secondary path, where under normal conditions dt is much shorter than dt s. Additionally, the signal C takes a time dt p to traverse the primary path. The arrangement of the loudspeakers in the Figures 2 and 3These are not representative; the path lengths of the sound waves 24 to the microphone 30 can be the same or different. In most cases, the loudspeakers 21 and 22 would be positioned next to each other. However, it is generally important to position the loudspeakers 21 and 22 in room 10 such that dt + dts > dtp for the time intervals dt + dts, so that the original sound cannot be canceled out at the location of the microphone 30. This occurs when the correction loudspeaker 22 and the location of the microphone 30 are very close together and the main loudspeaker 21 is very far away. However, this can also lead to good results. For example, if the microphone 30 and the correction loudspeaker 22 are positioned near the wall 13 furthest from the main loudspeaker 21, the entire desired signal N is canceled out there, and only there. This creates a room acoustic that suggests the absence of this back wall.All of this is only relevant if the correction speaker 22 and the main speaker 21 are separate speakers.

[0039] It should be noted that none of the times dt, dt s, or dt p are determined or known. Therefore, they are not included in the procedure. The time delay dt of the delayed playback of the correction speaker 22 corresponds to the time the user signal N requires to pass through the filter W*(z). This value is determined solely by the filter W*(z). No delay needs to be entered into the system or the procedure to intentionally send the signal Ñ- to the secondary path S at a later time.

[0040] Preferably, a common loudspeaker 23 is used, to which both signals N and Ñ- are superimposed, which were individually intended for the main loudspeaker 21 and the correction loudspeaker 22. This is certainly advantageous for cost reasons. In this case, the digital signal processor 60 can be arranged directly in the common loudspeaker 23, preferably integrated.

[0041] However, if a loudspeaker is already present and is to be reused, an additional correction loudspeaker 22 can be used. This is preferably a subwoofer, which preferably has acoustic properties similar to the existing loudspeaker, since it hardly needs to reproduce high frequencies. The digital signal processor 60 and possibly other components of the audio electronics 40 can then be arranged, preferably integrated, in this additional correction loudspeaker 22.

[0042] For the use of the procedure, a microphone 30 is no longer necessary. Preferably, it is now disconnected and removed. However, the loudspeaker(s) 21, 22, 23 must remain in their positions and are also connected to the signal processor 60 for use with the required audio electronics 40. Equivalent products for the audio electronics 40 should be used as those used in the characteristic measurements, with characteristics that are as similar as possible.

[0043] In this phase of the process, only the components of the signal processor 60 are used, as described in Fig. 3 and 6 as shown. Therefore, only the filter W*(z), which is connected upstream of the second or common output 63 or 64, is used. The lines to the other components in the signal processor 60, in particular to and from the LMS module, can therefore be interrupted.

[0044] When used in the signal processor 60, a user signal N is simultaneously routed to the first or common output 62, 64 and through the filter W*(z), and then as signal Ñ- to the second or common output 63, 64. Accordingly, signal N is played back from the main or common loudspeaker 21, 23 slightly earlier than signal Ñ- from the correction or common loudspeaker 22, 23, namely by the time dt required for the filter W*(z) to pass through.

[0045] During this time dt, room modes Ñ form in room 10, which are canceled out by the delayed playback of the correction loudspeaker 22. The original signal remains fully audible.

[0046] The process does not eliminate sounds originating from sources other than the digital user signal. The filter W*(z) can only react to and neutralize sound waves that have a causal relationship with the original signal and that are still present after the user signal N has passed through the filter.

[0047] Only the acoustic sound waves that have a causal relationship to the user signal N and are still present in room 10 after the time delay dt are canceled. These are the room modes Ñ.

[0048] Furthermore, the digital signal processor 60 according to the invention is described here using the Figures 7 and 8 described. These show the two variants of the signal processor 60 as a circuit diagram, for carrying out all process steps in each case.

[0049] A digital signal processor 60 according to the invention for use in a method described above comprises: a digital signal input 61 for supplying a digital output signal A, C or digital user signal N, either a first and a second output 62, 63 for connecting a main and a correction loudspeaker 21, 22, or a common output 64 for connecting a common loudspeaker 23, a microphone input 65 to which a microphone 30 can be connected for the characteristic measurements, an LMS module for carrying out algorithms with two inputs 71, 72 and a control output 73 for carrying out the characteristic measurements, wherein its first input 71 is connected to the digital signal input 61 and its second input 72 to the microphone input 65, wherein at least one filter position 74 is arranged before the first input 71 of the LMS module, which can be empty during the first characteristic measurement and occupied with an unchangeable filter S'*z during the second characteristic measurement, a filter position 70 for a changeable filter S'(z),W(z), which can be changed during the characteristic value measurements by the control output 73 of the LMS module and in which an unchangeable electronic filter W*(z) can be stored after completion of the second characteristic value measurement, wherein the filter position 70 is connected on the input side to the digital signal input 61 and is switchable on the output side by a first switch 77, so that on the output side it can be connected together with the microphone input 65 to a subtractor 75 and then to the second input 72 of the LMS module for the first characteristic value measurement, and can be connected to the second or common loudspeaker output 63, 64 for the second characteristic value measurement and for the use of the procedure in step b., as well as a connection from the digital signal input 61, which leads either to a second switch 78, which can optionally establish a connection to the first or second output 62, 63, or to the common output 64,so that for the first measurement of the characteristic value, the connection to the second or common output 63, 64 can be ensured, and for the second measurement of the characteristic value, as well as for the use of the procedure, the connection to the first or common output 62, 64 can be ensured.

[0050] Other, modified signal circuit diagrams are also suitable if the methods according to the invention can be carried out with them.

[0051] Fig. 7 shows the variant of the digital signal processor 60 with the first and second outputs 62, 63 for use with two separate main and correction loudspeakers 21, 22, while Fig. 8 The variant with the shared output 64 for the shared loudspeaker 23 is shown. Otherwise, the two versions are largely identical with only a few differences.

[0052] With these two variants of the signal processor 60 shown here, both characteristic value measurements and the application of the method can be carried out. For this purpose, a first switch 77 and, in the variant according to Fig. 7 , a second switch 78 is also provided. These allow the signal processor 60 to be configured for each use.

[0053] The first switch 77 is located at the output of the filter 70, which can be configured with S'(z), W(z), or W*(z), and can be connected either to the subtractor 75 or to the second or common output 63, 64. For the first measurement, the first switch 77 is connected to the subtractor 75 so that the signal from the output of the filter 70 is subtracted from the microphone signal B, which is input at the microphone input 65. Subsequently, for the second measurement and for the use of the procedure, the first switch 77 will route the signal from the filter 70 to the second or common output 63, 64.

[0054] In the second measurement, the subtractor 75 does not receive a second signal for subtraction and therefore passes the signal D unchanged to the LMS module. As is known, no microphone is used in this method. Therefore, an interrupter 79 can be provided after the microphone input 65 to prevent interference.

[0055] Additional interrupters 79 can be arranged before or after the filter station 74 and / or at the control output 73 of the LMS module. They can all interrupt the lines during operation of the procedure. However, they must maintain a connection during characteristic value measurements. The interrupters 79 are optional and can also be omitted.

[0056] Another switch 78, see Fig. 7This is only necessary if the signal processor has a first and a second output 62, 63. This switch 78 can route a signal A, C, N from the signal input 61 to either the first or the second output 62, 63. For the first characteristic measurement, the second switch 78 is connected to the second output 63 so that signal A can be routed to the correction loudspeaker 22 to determine its characteristics. Subsequently, for the second characteristic measurement and for using the procedure, the switch 78 is connected to the first output 62. In the second characteristic measurement, signal C, and in the use of the procedure, the user signal N, is routed to the second output 63.

[0057] As previously described, interrupters 79 can also be installed here, which can interrupt the connections to and from the LMS module. If these are not installed, the LMS module must be prevented by other means from influencing the unchangeable filter W*(z).

[0058] In summary, after the first measurement of the characteristic value, switch 77 and, if necessary, switch 78 must be switched, and after the second measurement of the characteristic value, the microphone can be removed and the LMS module can be decoupled with the upstream filter position 74.

[0059] As soon as the room or the position of the loudspeakers 21, 22, 23 is changed, the first and second characteristic measurements can be carried out again. For this, switches 77, 78 must be set accordingly again and the LMS module connected to the upstream filter station 74. In addition, microphone 30 must be set up and connected again.

[0060] In a preferred embodiment, the signal processor may include a tone generator 50 for performing the characteristic value measurements, wherein the tone generator may preferably generate white or pink noise.

[0061] According to the invention, a loudspeaker 21, 22, 23 is also described here, which comprises a digital signal processor 60 according to the invention, wherein it is preferably a subwoofer.

[0062] This loudspeaker is preferably the correction loudspeaker 22 or the common loudspeaker 23. Reference symbol list

[0063] 10 Room, for example recording studio 11 Seat of one person 12 Additional speakers, optional 13 Wall of the room 21 Main speaker 22 Correction speaker 23 Common speaker 24 Sound waves 30 Microphone, microphone location 40 Audio electronics, general 41 Digital-to-analog converters 42 Preamplifiers 43 Power amplifiers 44 Microphone amplifiers 45 Analog-to-digital converters 46 Cables 50-tone generator 60 Digital signal processor 61 Signal input for a digital signal 62 First output for the main speaker 63 Second output for the correction speaker 64 Output for the common speaker 65 Microphone input 70 Filter slot, for variable or fixed filter 71 First input to the LMS module 72 Second input to the LMS module 73 Control output from the LMS module to the variable filter 74 Filter slot 75 Subtractor for calculating the difference when combining two signals 77 First switch 78 Second switch 79 Interrupter for temporary disconnection, optional A Digital output signal, during the first characteristic measurement B Digital end signal, during the first characteristic measurement C Digital output signal, during the second characteristic measurement D Digital end signal, during the second characteristic measurement N Digital user signal for using the method Ñ Filtered user signal, for eliminating room modes Ñ Room mode, which forms in the room due to emitted sound waves LMSLMS Module for carrying out numerical gradient methods SS Secondary path S(z) First transfer function on the secondary path S'(z) Electronic filter, modifiable and storable S'*(z) Stored electronic filter S'(z), unmodifiable PPrimary path P(z)second transfer function on the primary path W(z)electronic filter, modifiable and storable W*(z)stored electronic filter W(z), unmodifiable dt Time delay in the electronic filter W(z) dt p Travel time of the sound wave in the primary path P dt s Travel time of the sound wave in the secondary path S

Claims

1. A method for eliminating room modes (N) which form as resonances in a room (10) when a digital user signal (N) is played by a main loudspeaker (21), characterized by the following steps: a. establishment and performing of characteristic value measurements, by i. positioning a main loudspeaker (21) and a correction loudspeaker (22) in a room (10), for example in a recording studio, wherein these loudspeakers (21, 22) can be two separate or one common loudspeaker (23); positioning a microphone (30) in this room (10); providing a digital signal processor (60) with a signal input (61) for inputting and processing digital signals (A, C, N), a first (62) and a second (63) loudspeaker output for the main loudspeaker (21) and the correction loudspeaker (22), which can be combined to form a common loudspeaker output (64) for the common loudspeaker (23), and a microphone input (65) for the microphone (30), connecting the signal processor (60), the one or more loudspeaker(s) (21, 22, 23) and the microphone (30) to sound electronics (40), preferably comprising at least one digital-to-analog converter (41), power amplifier (43), microphone amplifier (44), analog-to-digital converter (45) and cable (46), for generating and capturing sound waves (24) by means of loudspeaker (21, 22, 23) and microphone (30); ii. performing a first characteristic measurement, in which a first transfer function (S(z)), which represents the change in a digital signal (A), after it has been picked-up on a secondary path (S), which is played via the correction loudspeaker (22) or the common loudspeaker (23) as sound waves (24), at the microphone (30) and captured as a digital signal B [AS(z)=B], is reproduced by a variable electronic filter (S'(z)), using an LMS (Least Mean Square) module for performing a numerical gradient method, preferably using the filtered-x LMS method, which generates the variable filter (S'(z)), based on the knowledge of the original signal (A), and adapts it until the original signal (A), after it has passed through this filter (S'(z)), corresponds to the captured signal (B) at the end of the secondary path (S) and optionally cancels [AS(z) - AS'(z) ≈ 0] it accordingly at an electronic subtractor (75); iii. saving this electronic filter (S'*(z)), which thus becomes unchangeable; iv. performing a second characteristic measurement, in which a second transfer function (P(z)), which represents the change in a digital signal C, after, on a primary path (P), which is played as sound waves (24) via the main loudspeaker (21) or the common loudspeaker (23), it has been picked up at the microphone (30) and captured [CP(z)=D] as a digital signal (D), it is partially reproduced by a variable electronic filter (W(z)), using the LMS module for performing a numerical gradient method, preferably using the filtered-x LMS method, which generates the variable filter (W(z)), based on the knowledge of the original signal (C), and adapts it until the original signal (C), after it has passed through the stored filter (S'*(z)), when it passes through this variable filter (W(z)) and subsequently the secondary path (S), corresponds as closely as possible to the negative of the originally captured signal (D) from the primary path P and minimizes [CP(z) - CWS(z) ≈ 0] this accordingly when merging at the microphone (30); v. saving the electronic filter (W*(z)), which thus becomes unchangeable; b. setting up and using the method with a user signal (N), by i. positioning the main loudspeaker (21) and the correction loudspeaker (22), or the common loudspeaker (23), at the same locations in the same room (10) as in the characteristic value measurements, with the same sound electronics (40), required for this purpose, and connecting them to the digital signal processor (60) as in step a.; ii. directing the digital user signal (N) to the first or common output (62, 64) and playing it through the main or common loudspeaker (21, 23), wherein room modes (N) are formed in the room (10); iii. simultaneously directing this user signal (N) through the last stored filter (W*(z)) in the signal processor (60) and subsequently forwarding it to the second or common output (63, 64), and because of the time (dt) required by the filter (W*(z)), delayed playing of this filtered user signal (N-) through the correction or common loudspeaker (22, 23), iv. whereby the room modes (N) of the digital user signal (N) which are still present in the room (10) after the time delay (dt) are deleted.

2. The method according to claim 1, characterized in that, before step a.iv), the digital signal processor (60) is set up for the second characteristic value measurement by directing the output of the variable filter (W(z)) to the second or common output (63, 64) to the correction loudspeaker (22) or to the common loudspeaker (23).

3. The method according to any one of the preceding claims, characterized in that the microphone (30) is positioned, in step a., at a location where a person (11) is expected to be in step b., or near a wall (13) of the room (10) which is far away from the main loudspeaker (21).

4. The method according to any one of the preceding claims, characterized in that the room (10) is between 10 and 100 m2 in size.

5. The method according to any one of the preceding claims, characterized in that the time delay (dt) of the delayed playing of the correction loudspeaker (22) corresponds to the time which the user signal (N) requires to pass through the filter W*(z).

6. The method according to any one of the preceding claims, characterized in that the main loudspeaker (21) and the correction loudspeaker (22) are separate loudspeakers.

7. The method according to claim 6, characterized in that, in step a., the correction loudspeaker (22) is positioned at a location in the room (10) such that a sound wave (24), emitted by the main loudspeaker (21), arrives at the microphone (30) earlier than a sound wave (24), emitted later by the correction loudspeaker (22) with the time delay (dt).

8. The method according to any one of the preceding claims, characterized in that a common loudspeaker (23) is used, to which the two signals, which were individually provided for the main loudspeaker (21) and the correction loudspeaker (22), are fed in a superimposed manner.

9. The method according to any one of the preceding claims, characterized in that, in step b., no microphone (30) is used wherein the microphone (30) is preferably disconnected before step b. of the method.

10. The method according to any one of the preceding claims, characterized in that the main loudspeaker (21) and / or the correction loudspeaker (22), or optionally the common loudspeaker (23) are subwoofers.

11. The method according to any one of the preceding claims, characterized in that the numerical gradient method is a filtered-x LMS algorithm.

12. A digital signal processor (60) for use in a method according to any one of the preceding claims, comprising - a digital signal input (61) for feeding in a digital output signal (A, C) or digital user signal (N), - either a first and a second output (62, 63) for connecting a main and a correction loudspeaker (21, 22), or a common output (64) for connecting a common loudspeaker (23), - a microphone input (65) to which a microphone (30) can be connected for the characteristic value measurements, - an LMS module for performing algorithms with two inputs (71, 72) and a control output (73) for performing the characteristic value measurements, wherein its first input (71) is connected to the digital signal input (61) and its second input (72) is connected to the microphone input (65), - wherein a filter position (74) is disposed in front of the first input (71) of the LMS module, which is empty during the first characteristic value measurement and is occupied by an unchangeable filter (S'*(z)) during the second characteristic value measurement, - a filter position (70) for a variable filter (S'(z), W(z)), which can be changed during the characteristic value measurements by the control output (73) of the LMS module and in which a fixed electronic filter (W*(z)) can be stored after completion of the second characteristic value measurement, wherein the filter position (70) is connected on the input side to the digital signal input (61) and is switchable on the output side by a first switch (77), such that, on the output side, for the first characteristic value measurement, it can be led, together with the microphone input (65), to a subtractor (75) and subsequently to the second input (72) of the LMS module, and for the second characteristic value measurement as well as for the use of the method in step b. is connected to the second or common loudspeaker output (63, 64), - as well as a connection from the digital signal input (61), which leads either to a second switch (78), which can optionally establish a connection to the first or second output (62, 63), or to the common output (64), such that the connection to the second or common output (63, 64) is ensured for the first characteristic value measurement and the connection to the first or common output (62, 64) is ensured for the second characteristic value measurement and for the use of the method.

13. The digital signal processor according to claim 12, characterized in that one or more interrupters (79) are disposed which, for the use of the method after completion of the characteristic value measurements, can interrupt the connection to the first and / or second input (71, 72) of the LMS module and / or the control connection from the LMS module to the filter position (70).

14. The digital signal processor according to claim 12 or 13, characterized in that it comprises a tone generator (50) for performing the characteristic value measurements, wherein the tone generator (50) can preferably generate pink noise.

15. A loudspeaker (21, 22, 23), comprising a digital signal processor (60) according to any one of claims 12 to 14, wherein it is preferably a subwoofer.

16. The loudspeaker (22, 23) according to claim 15, wherein the loudspeaker (22, 23) is a correction loudspeaker (22) or a common loudspeaker (23).

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