Infotainment system and method for operating an infotainment system

Abstract

The invention relates to an infotainment system, comprising an audio playback system which comprises more than two loudspeakers (102), and an audio capture device which has a microphone (109). The microphone (109) emits a microphone signal (113) which contains a voice signal and audio signals of the audio playback system. The infotainment system further comprises an up-mix stage (301), which generates, from an input signal (104) having M channels, an intermediate signal (303) having L channels and an output signal (201) having N channels, each of which is supplied to a loudspeaker (102) of the audio playback system, where L < N. A compensation stage (112), to which the microphone signal (113) is supplied as a useful signal and which has a filter (118), is supplied with a reference signal (119) in order to generate a filtered signal (D) from the reference signal (119). The compensation stage (112) is designed to extract the filtered signal (D) from the useful signal (113). The removal of undesired signals is thus less complex than in conventional infotainment systems, because an intermediate signal having fewer channels than the output signal (201) is used as the reference signal (119) for the compensation stage (112).

Description

[0001] Sennheiser electronic SE & Co. KG

[0002] Am Labor 1, 30900 Wedemark

[0003] Infotainment system and operating procedures for an infotainment system

[0004] Field of invention

[0005] The invention relates to an infotainment system that combines a media playback system, in particular an audio playback system, with a telecommunications system or with a voice control device. The infotainment system is characterized by improved properties of an audio acquisition device for speech signals that interacts with the telecommunications system or with the voice control device.

[0006] background

[0007] Generally, an audio playback system of a high-performance infotainment system comprises multiple loudspeakers, each driven by an audio playback channel to create surround sound or spatial sound. In addition to the loudspeakers, the media playback system may also include screens for displaying video signals, but this is not relevant to the present invention. Therefore, the following refers only to an audio playback system (hereinafter referred to as "audio system") without limitation of generality.

[0008] The infotainment system also includes a telecommunications system or a voice control device that interacts with an audio receiver for voice signals. The audio receiver is equipped with a microphone for capturing audio signals, particularly voice signals from a user of the infotainment system. These voice signals can be part of a conversation the user is conducting via the telecommunications system. In this case, the audio receiver is also referred to as a hands-free system. The voice signals can also be voice commands for voice control, which are then processed by an associated control unit, such as a voice assistant. Furthermore, the infotainment system may include a camera and a...

[0009] *20250661348* The invention includes a screen for capturing and / or playing back video signals, as is required, for example, for video conferencing. However, the capture and playback of video signals is not further relevant to the present invention; nevertheless, the present invention can also be applied in infotainment systems that capture and play back video signals in addition to audio signals.

[0010] Regardless of whether the speech signals are part of a conversation or voice commands, it is desirable that the speech signals captured by the microphone of the audio capture device are not superimposed on other audio signals.

[0011] In all infotainment systems where speech signals are simultaneously captured by the microphone of the audio pickup unit during audio playback, the problem arises that the speech signals are superimposed on the audio signal playback, impairing the intelligibility of the speech signals, regardless of whether the speech signals are contributions to a telephone conversation or voice commands to a voice assistant. This problem also occurs, albeit in a lesser form, when a microphone array is used instead of a microphone with or without directional characteristics. For illustrative purposes, it is assumed that the audio signal playback refers to music signals, as distinct from the speech signals captured by the microphone. This is explained below using the example of an infotainment system in a vehicle. If the infotainment system's audio system outputs stereo signals, i.e.,Since the system only reproduces two output channels, the described problem can be solved with relatively little effort using a 2-channel adaptive filter, where the loudspeaker signals serve as reference signals for the adaptive filter to remove the music signal component from the microphone signal of the audio capture device. Such multi-channel adaptive filters are known in the art and are used, for example, for acoustic echo cancellation (AEC) in hands-free devices.With an increasing number of output channels, the conventional approach described above has disadvantages, which is relevant, for example, in modern motor vehicles equipped with multi-channel audio playback systems. In such systems, a stereo input signal is converted by an upmixer to provide multiple audio output channels, which are then played back through a corresponding number of loudspeakers to create surround sound. In one specific implementation, 14 output channels for different loudspeakers at various locations within the vehicle's interior can be generated from a single stereo input signal.Generally speaking, an upmixer can generate N output signals from M input signals, where, for example, M = 1 represents a mono input signal and M = 2 a stereo input signal, and where N is greater than M (N > M) and usually corresponds to the number of installed loudspeakers. It is fundamentally possible to remove the music signal generated from the N output signals from the microphone signal using a multi-channel adaptive filter. This is based on the idea that every audio signal emitted by an audio source and picked up by a microphone has a transfer function that depends, among other things, on the acoustic path—that is, the relative position of the audio source to the microphone—and the acoustic properties of the room in which the audio signals are emitted and received.The multi-channel adaptive filter essentially determines the N acoustic paths from the loudspeakers to the microphone of the hands-free device in order to filter the N loudspeaker signals with the corresponding estimated transfer functions and finally subtract them from the sum of the signals picked up by the microphone to remove the music signals from the microphone signal. Disadvantages already arise with N > 1, and these increase as N becomes larger. These disadvantages include:

[0012] (a) The computational effort for filtering and adjusting the adaptive filter increases approximately linearly with N and becomes very large for large N.

[0013] (b) Due to the nature of the upmixing process, the mutual correlation of the N loudspeaker signals is high, so the adaptation of the filters suffers from the so-called ambiguity problem, which is described for two reference channels in the article by M.M. Sondhi et al. (M.M. Sondhi, Dr. Morgan, and J.L. Hall. Stereophony acoustic echo cancellation—an overview of the fundamental problem. IEEE Signal Processing Letters, 2(8):148–151, August 1995). In particular, the loudspeaker-microphone transfer functions derived from the optimization of the adaptive filter are not unique, which creates the risk that an adaptation error will increase abruptly even with a small change in the actual acoustic transfer functions. Such small changes can be caused, for example, by the movement of occupants in a vehicle.

[0014] (c) The large number of filter coefficients that need to be determined makes the adaptation of the filter slow.

[0015] It should be noted here that if the upmixing process consisted only of linear time-invariant operations, then it would suffice to use an M-channel adaptive filter and the original M-channel music signal as the reference signal. This would overcome the aforementioned disadvantages because, in practical applications, M ≤ N. In the case of purely linear time-invariant operations, the upmixing parameters would be implicitly included in the transfer function estimates of the adaptive filter.

[0016] In reality, typical upmixing algorithms are based on a short-term analysis of the input signals and a corresponding short-term adaptation of the upmix parameters with block sizes on the order of milliseconds (Carlos Avendano and Jean-Marc Jot. Frequency domain techniques for stereo to multichannel upmix. In Audio Engineering Society Conference: 22nd International Conference: Virtual, Synthetic, and Entertainment Audio, June 2002). This highly time-dependent behavior, which can even exhibit nonlinear components, prevents the adaptive filter from converging because it cannot follow the changes in the upmix parameters quickly enough. A non-converging adaptive filter results in the reference signal components not being removed from the desired signal, or at least not completely. In the example described, the music signal would remain audible or at least cause interference in the microphone signal.

[0017] Starting from this, it is an object of the present invention to create an infotainment system in which music signals emitted by an audio playback system are removed more effectively and efficiently from a microphone signal of an audio recording device than is currently possible with known technical solutions.

[0018] In the priority-establishing German patent application, the German Patent and Trademark Office searched the following documents: US 2019 / 0 373 390 A1, US 2020 / 0 136 675 A1 and WO 2014 / 182 478 A1.

[0019] According to a first aspect, an infotainment system is proposed. The infotainment system comprises an audio playback system with more than two loudspeakers and an audio capture device with a microphone. The microphone captures voice signals from a user of the infotainment system as well as audio signals from the audio playback system and outputs these signals as a microphone signal containing the voice signal and the audio signals from the audio playback system. The infotainment system further comprises an upmix stage that generates an intermediate signal with L channels from an input signal with M channels, and from the intermediate signal an output signal with N channels, each of which is fed to a loudspeaker of the audio playback system, where M, L, and N are natural numbers for which L < N.The infotainment system further includes a compensation stage to which the microphone signal is fed as the useful signal and which has a filter to which the intermediate signal is fed as a reference signal in order to generate a filtered signal from the reference signal, which replicates the reference signal in a form as it is contained in the useful signal, with the compensation stage being set up to subtract the filtered signal from the useful signal.

[0020] The infotainment system has the advantage of being less complex than conventional infotainment systems designed to remove music signals from a speech signal because it uses an intermediate signal with fewer channels than the output signal as a reference signal for the compensation stage. This avoids the difficulties described earlier that arise with compensation stages when the reference signal has numerous channels.

[0021] According to a beneficial further development, the upmix stage comprises a first and a second component. The first component is designed to apply at least partially time-varying and / or nonlinear operations to an input signal with M channels in order to generate the intermediate signal with L channels.

[0022] In an advantageous further development, the second component of the upmix stage is designed to generate the output signal with N channels from the intermediate signal with L channels through linear operations.

[0023] In a suitable further training, the second component of the upmix stage is set up to generate the output signal with N channels from the intermediate signal with L channels through time-invariant and linear operations.

[0024] The two components of the upmix stage make it possible to separate time-variant and / or non-linear operations from time-invariant and / or linear operations, with the result that, once the intermediate signal is available, only time-invariant and / or linear operations need to be performed to obtain the output signal for the loudspeakers.

[0025] Advantageously, the filter in the compensation stage is designed as an adaptive filter. The adaptive filter can dynamically adjust the filter parameters to changing transfer functions, which can be caused, for example, by changes in passenger seating positions in a vehicle with an integrated infotainment system. However, this advantage of adaptability to changing transfer functions is not limited to infotainment systems in vehicles.

[0026] In one embodiment, the input signal with M channels contains fewer channels than the output signal with N channels, such that M < N. This allows the upmix stage to take advantage of a larger number of N loudspeakers when only a smaller number of M channels are available in the input signal.

[0027] According to a second aspect of the invention, an operating method for an infotainment system is proposed. The infotainment system includes an audio playback system comprising more than two loudspeakers and an audio acquisition device with a microphone that acquires speech signals from a user of the infotainment system as well as audio signals from the audio playback system and outputs them as a microphone signal. The infotainment system further includes an upmixing stage that generates an output signal with N channels from an input signal with M channels, and a compensation stage comprising a filter. The method comprises generating an intermediate signal with L channels from the input signal with M channels, generating the output signal with N channels from the intermediate signal with L channels,

[0028] The output signal is sent via N channels to the loudspeakers of the audio playback system. The microphone signal is fed to the compensation stage as the desired signal. The intermediate signal is fed to the filter as a reference signal, and a filtered signal is generated from the reference signal. This filtered signal replicates the reference signal in the form it is contained in the desired signal. The filtered signal is then subtracted from the desired signal, where M, L, and N are natural numbers for which L < N. This operating method makes it possible to realize the advantages already mentioned in connection with the infotainment system.

[0029] According to an advantageous further development of the operating procedure, the intermediate signal is generated with L channels from the input signal with M channels by applying at least partially time-variant and / or non-linear operations.

[0030] Advantageously, the generation of the output signal with N channels from the intermediate signal with L channels can be achieved by applying time-invariant and linear operations.

[0031] The fact that only time-invariant and / or linear operations are applied to the intermediate signal to generate the output signals ensures improved removal of unwanted music signals from a speech signal captured by an audio capture device, compared to conventional infotainment systems.

[0032] According to a third aspect of the invention, a computer program with instructions is proposed which, when the computer program is executed by a processor, performs an operating method according to the second aspect of the invention.

[0033] Brief description of the characters

[0034] The invention is explained in more detail below using one embodiment as an example, with reference to the accompanying figures. All figures are purely schematic and not to scale. They show:

[0035] Fig. 1 shows a schematic representation of an infotainment system;

[0036] Fig. 2 shows a schematic representation of an infotainment system that is not part of the invention;

[0037] Fig. 3 shows a schematic flowchart of an upmix algorithm with 2 components;

[0038] Fig. 4A is a schematic representation of an infotainment system according to the invention;

[0039] Fig. 4B shows a section of Fig. 4A with the compensation stage in greater detail; and Fig. 5 shows a schematic flowchart for an operating procedure for an infotainment system according to the invention.

[0040] Identical or similar elements in the figures are marked with the same or similar reference symbols.

[0041] Fig. 1 shows a schematic representation of an infotainment system 100, which includes a media playback system in the form of an audio playback system (audio system), which is shown in a highly simplified manner in Fig. 1. Of the components of the audio system, only an upmix stage 101 and loudspeakers 102a-h are shown in Fig. 1, each of which is individually connected to the upmix stage 101 via electrical lines 103. The upmix stage 101 receives a stereo input signal 104, from which the upmix stage 101 generates 8 output channels (N = 8) in the illustrated embodiment, so that each output channel is available for playback in one of the 8 loudspeakers 102a-h. The number of 8 loudspeakers is chosen only as an example, and other embodiments may have more or fewer than 8 loudspeakers. In Fig.Figure 1 shows the infotainment system 100 installed in a vehicle (not shown), the interior of which is symbolized by a line 106. The interior contains a driver's seat 107a, a front passenger seat 107b, and a rear bench seat 107c, with only the driver's seat 107a occupied by a driver 108. A telecommunications system belonging to the infotainment system 100 is also installed in the vehicle interior 106. The telecommunications system has one or more microphones designed to capture voice signals from passengers in the vehicle. For clarity, Figure 1 shows only a single microphone 109 of the telecommunications system, designed to capture voice signals 110 from the driver 108. When the driver 108 makes a telephone call using the microphone 109, the audio signal of the other party is reproduced, for example, as a mono signal via headrest speakers 111a, b.In other embodiments, other loudspeakers arranged in the interior 106 of the vehicle are used to reproduce the audio signal of a conversation partner, whereby reproduction using a single loudspeaker is also possible.

[0042] Naturally, microphone 109 also records the audio signals emitted by the headrest speakers 111 a, b, so that without countermeasures a disturbing echo effect would occur in a telephone conversation. The echo effect is eliminated by a compensation stage 112. For this purpose, compensation stage 112 includes an adaptive filter that receives a microphone signal 113 from microphone 109 as the input signal and a loudspeaker signal 114, which drives the headrest speakers 111 a, b, as the reference signal. Based on these two signals 113, 114, compensation stage 112 generates an output signal 116 in which the loudspeaker signal 114 from the headrest speakers 111 a, b is removed from the microphone signal 113 of microphone 109.The loudspeaker signal 114 can be removed by the adaptive filter of compensation stage 112 without significant computational effort because the transfer function of the headrest loudspeakers 111a,b is linear and time-invariant. If the audio system is playing music via the loudspeakers 102a-h of the infotainment system 100 while the driver is on a telephone call using the hands-free system of the telecommunications system, the additional problem arises that the microphone 109 picks up not only the driver's voice signal 110 108, but also the music being played. This is often perceived as annoying by the person on the other end of the call. This unwanted "crosstalk" of the music playback from the loudspeakers 102a-h is shown in Fig.

[0043] Figure 1 illustrates this with arrows 117. Different upmixing functions and propagation delays of the playback signals from the individual loudspeakers 102a-h result in different transfer functions of the sound signals from the loudspeakers 102a-h to the microphone 109, which is also illustrated in Figure 1 by the different lengths of arrows 117. To remove the music signal from the microphone signal 113, it is necessary to feed reference signals 119, which are described in detail below, to a filter 118 in the compensation stage 112. In a preferred embodiment, the filter 118 is an adaptive filter.

[0044] Fig. 2 shows a schematic block diagram of a highly simplified infotainment system 200, which is not part of the invention. The infotainment system 200 illustrates how the described problem is conventionally solved. As already described in connection with Fig. 1, the microphone 109 captures the music signals from the loudspeakers 102a-h in addition to the driver's speech signal 110. The output signal 113 of the microphone 109 is therefore a superposition of the driver's speech signal 110 and the music signals. The echo cancellation known as Acoustic Echo Cancellation (AEC) is not relevant here because the echo signal is independent of the music signal and therefore has no influence on the removal of the music signal from the microphone signal 113. Music playback in the system shown in Fig.

[0045] In the embodiment shown in Figure 2, an input signal with M channels is used, where M = 1 (mono signal) or M = 2 (stereo signal). In the upmix stage 101, N output channels 201 are generated from this signal, where N ≤ M, for example, N = 14 and typically N = n. The entirety of all output channels is also designated by the reference numeral 201. Each output channel 201 is fed to one of the loudspeakers 102a-n. The output channels 201 are also provided as reference signals 119 for the compensation stage 112. The compensation stage 112 receives the microphone signal 113 as the desired signal and outputs the output signal 116, from which the music playback has been removed.However, this approach suffers from the aforementioned disadvantages, namely that the computational effort is high due to the numerous output channels, and that small changes to the actual transfer functions result in large mismatches in the compensation stage 112, which cause audible disturbances.

[0046] As explained in the introduction, typical upmix algorithms that generate a plurality of N output channels from an input signal with one or two input channels are based on a short-term analysis of the input signal and a corresponding short-term adaptation of the upmix parameters. They exhibit highly time-dependent behavior, which can even include nonlinear components. Therefore, the adaptive filter does not converge because it cannot follow the changes in the upmix parameters quickly enough or replicate them linearly.

[0047] Fig. 3 shows a schematic block diagram for an alternative upmix algorithm, shown as a whole as block 301, which generates N output channels 201 from an input signal 104 with M channels. Overall, the upmix algorithm 301 features time-variant and non-linear operations and decomposes the upmixing process into two stages performed by different components. The proposed alternative upmix algorithm is described, for example, in “Sebastian Kraft and Udo Zölzer. Stereo signal separation and upmixing by mid-side decomposition in the frequency-domain. 2015” and features a first component 302 that generates a number of L channels of an intermediate signal 303 by means of time-variant operations, which can also be non-linear, where L > M can optionally be.In a second component 304 of the upmix algorithm 301, the L channels of the intermediate signal 303 are converted into N output channels 201 by means of linear and time-invariant operations, where L < N, for example, M = 2, L = 4, and N = 5. In other embodiments, the difference between L and N is greater, i.e., L ≤ N. In a practical embodiment, a configuration with the parameters M = 2, L = 5, and N = 13 was successfully tested. Since the N output channels 201 are derived from the L channels of the intermediate signal 303 by purely linear time-invariant operations, it is proposed according to the invention to use the L channels of the intermediate signal 303 as a reference signal 119 for the compensation stage 112, because the upmix parameters are implicitly included in the transfer function estimates of the adaptive filter of the compensation stage. However, the invention is not limited to M = 1 or M = 2.Rather, M could also be 3, 4, or even larger, for example M = 5, as would be the case with a surround signal as input signal 104. It is important for the effective application of the technical teaching of the present invention that N > M holds true.

[0048] Optionally (for example, due to technically unavoidable deviations in an implementation), the second component 304 of the upmix algorithm could exhibit small components that are not completely linear and time-invariant. If the transfer functions of the second component 304 change only slowly, the adaptive filter can partially compensate for such deviations. In an optional embodiment of the invention, the generation of the output signal 201 with N channels from the intermediate signal 303 with L channels can be achieved exclusively through time-invariant, linear operations in order to simplify the adaptation of the filter 118 as much as possible.

[0049] The term component 302,304 refers both to the respective part of algorithm 301 and to the hardware on which the respective part of the algorithm is executed.

[0050] To further illustrate this, consider a simple application example where, in a first step, a stereo signal is decomposed using a nonlinear and time-varying "Primary Ambient Signal Decomposition" (PAD) process into a center signal, which is panned in both channels, and two additional "ambience" signals. The "ambience" signals represent the uncorrelated components between the stereo channels. In this case, there are therefore L = 3 intermediate signals: the center signal and the two ambience signals, generated using time-varying and nonlinear operations. An example of the application of a nonlinear operation is an audio compressor, which compresses an input signal according to a nonlinear characteristic curve. For the sake of completeness, it should be noted that a linear operation can also be understood as the application of a gain factor of 1.In a second step, the center signal is distributed to, for example, three front speakers using amplitude panning, and the ambience signals are distributed to all speakers (e.g., N = 13) by applying equalizers and fixed but distinct all-pass filters. In other words, N = 13 output signals are generated from the L = 3 intermediate signals using linear time-invariant operations. The first and second steps of the PAD method correspond to the first and second components 302 and 304, respectively, of algorithm 301 from Fig. 3.

[0051] Figure 4A shows a schematic block diagram of an infotainment system 400, in which the upmix algorithm 301 described with reference to Figure 3 is used. An input signal 104 with M channels is fed into the upmix algorithm 301. The first component 302 of the upmix algorithm 301 generates an intermediate signal 303 with L channels using time-variant and, if necessary, non-linear operations. A second component 304 generates N output channels 201 from the intermediate signal 303 based on linear and time-invariant operations. These output channels are fed to a corresponding number of loudspeakers 102a-n (where n = N) for playback.The L channels of the intermediate signal 303 are also supplied as reference signals to the compensation stage 112, which receives the microphone signal 113 as the useful signal and outputs a filtered signal 116 as the output signal, from which signal components corresponding to the reference signals are removed in order to improve the intelligibility of the speech signals captured by the microphone 109.

[0052] In another embodiment, the microphone 109 belongs to a voice assistant that captures and executes voice commands from a user of the voice assistant. In this embodiment as well, it is advantageous that other audio signals, for example, music that is played simultaneously during voice input, are filtered out of the microphone signal in order to improve the recognition of the voice commands.

[0053] Fig. 4B shows a section of Fig. 4A with the compensation stage 112 in greater detail. The microphone signal 113 is fed as the useful signal to an adder 401 as the first input signal, and the reference signal 119 is fed to the adaptive filter 118. An output signal D of the adaptive filter 118 forms a second input signal for the adder 401 with the opposite sign. The output signal Y of the adder 401 is fed to the adaptive filter 118 as an error signal E. The adaptive filter 118 attempts to minimize the error signal E by adjusting the filter coefficients. In this way, the compensation stage 112 reduces, completely or at least partially, signal components in the useful signal that are correlated with the reference signal. Finally, a schematic flowchart for an operating procedure for an infotainment system according to the invention is shown. In a first step S1, an intermediate signal with L channels is generated from an input signal with M channels.In step S2, an output signal with N channels is generated from the intermediate signal with L channels. In step S3, the output signal with N channels is sent to the loudspeakers of the audio playback system belonging to the Infotainment System 400. In step S4, the microphone signal 113 is fed to the compensation stage 112 as the desired signal. In step S5, a reference signal is fed to the filter 118 in the compensation stage 112, from which a filtered signal is generated that replicates the reference signal in a form as it is contained in the desired signal. Finally, in step S6, the filtered signal is subtracted from the desired signal.

[0054] Individual components or functionalities of the present disclosure are described as software or hardware solutions in the exemplary embodiments. However, this does not mean that a functionality described as a software solution cannot also be implemented in hardware, and vice versa. Likewise, hybrid solutions are conceivable for those skilled in the art, in which components and functionalities are partially implemented simultaneously in software and hardware.

[0055] In the claims, the word "comprise" does not include any further elements or steps, and the indefinite article "a" does not exclude a plurality.

[0056] A single unit or device can perform the functions of several features mentioned in the claims. The fact that individual functions and elements are listed in different dependent claims does not mean that a combination of these functions and elements cannot be used advantageously.

[0057] References

[0058] [1] Herbert Buchner, Jacob Benesty, and Walter Kellermann. Multichannel Frequency-Domain Adaptive Filtering with Application to Multichannel Acoustic Echo Cancellation, pages 95-128. Springer Berlin Heidelberg, Berlin, Heidelberg, 2003. Reference symbol list

[0059] Infotainment system

[0060] Upmix - Stage ah Speaker

[0061] lines

[0062] Stereo input signal

[0063] Interior AC seats

[0064] driver

[0065] microphone

[0066] Speech signal a, b Headrest speaker

[0067] Compensation level

[0068] Microphone output signal, headrest speaker signal

[0069] Output signal compensation stage

[0070] Arrows

[0071] Adaptive Filter

[0072] Reference signals music infotainment system to speakers

[0073] Upmix algorithm

[0074] First component

[0075] Intermediate signal

[0076] Second component

[0077] I nfotai n me ntsy ste m adder

Claims

Claims 1. Infotainment system with an audio playback system comprising more than two loudspeakers (102) and with an audio acquisition device having a microphone (109) that captures speech signals from a user of the infotainment system as well as audio signals from the audio playback system and outputs them as a microphone signal (113) containing the speech signal and the audio signals of the audio playback system, with an upmix stage (301) that generates an intermediate signal (303) with L channels from an input signal (104) with M channels and an output signal (201) with N channels from the intermediate signal (303), each of which is supplied to a loudspeaker (102) of the audio playback system, where M, L and N are natural numbers for which L < N, with a compensation stage (112) to which the microphone signal (113) is supplied as the useful signal and which has a filter (118) to which the intermediate signal (303) is supplied as a reference signal (119). suppliedto generate a filtered signal (D) from the reference signal (119) which replicates the reference signal in a form as it is contained in the useful signal, wherein the compensation stage (112) is configured to subtract the filtered signal (D) from the useful signal (113).

2. Infotainment system according to claim 1, wherein the upmix stage (301) comprises a first and a second component (302, 304), wherein the first component (302) is configured to apply at least partially time-variant and / or non-linear operations to an input signal (104) with M channels in order to generate the intermediate signal (303) with L channels.

3. Infotainment system according to claim 1 or 2, wherein the upmix stage (301) comprises a first and a second component (302, 304), wherein the second component (304) of the upmix stage (301) is configured to generate the output signal (201) with N channels from the intermediate signal (303) with L channels by linear operations.

4. Infotainment system according to claim 3, wherein the second component (304) of the upmix stage (301) is configured to generate the output signal (201) with N channels from the intermediate signal (303) with L channels by time-invariant and linear operations.

5. Infotainment system according to one of the preceding claims, wherein the filter is designed as an adaptive filter (118).

6. Infotainment system according to one of the preceding claims, wherein the input signal (104) with M channels contains fewer channels than the output signal (201) with N channels, such that M < N.

7. Operating method for an infotainment system with an audio playback system comprising more than two loudspeakers (102) and with an audio acquisition device comprising a microphone (109) that acquires speech signals from a user of the infotainment system as well as audio signals from the audio playback system and outputs them as a microphone signal (113), and with an upmix stage (301) that generates an output signal (201) with N channels from an input signal (104) with M channels, and a compensation stage (112) comprising a filter (118), wherein the method comprises generating an intermediate signal (303) with L channels (S1) from the input signal (104) with M channels, and generating the output signal (201) with N channels (S2) from the intermediate signal (303) with L channels. Output (S3) of the output signal (201) with N channels to the loudspeakers (102) of the audio playback system, feed the microphone signal (113) as the useful signal to the compensation stage (112) (S4), feed the intermediate signal (303) as the reference signal (119) to the filter (118) and generate a filtered signal (D) from the reference signal (S5) which replicates the reference signal in a form as it is contained in the useful signal (113), and subtract the filtered signal (D) from the useful signal (113) (S6), where M, L and N are natural numbers for which L < N.

8. Operating method according to claim 7, wherein the generation of the intermediate signal (303) with L channels from the input signal (104) with M channels is carried out by applying at least partially time-variant and / or non-linear operations.

9. Operating method according to claim 7 or 8, wherein the generation of the output signal (201) with N channels from the intermediate signal (303) with L channels is carried out by applying linear operations.

10. Operating method according to claim 9, wherein the generation of the output signal (201) with N channels from the intermediate signal (303) with L channels is carried out by applying time-invariant and linear operations.

11. Operating method according to any one of claims 7 to 10, wherein the input signal (104) with M channels contains fewer channels than the output signal (201) with N channels such that M < N.

12. Computer program with instructions which, when the computer program is executed by a processor, perform a method according to any one of claims 7 to 11.

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