Description

The present invention relates to audio reproduction, and, in particular, to an apparatus and a method for automatic adaption of a loudspeaker to a listening environment.

A general issue in audio reproduction with loudspeakers is that during sound reproduction the loudspeaker is interacting with its environment, which is often an enclosed space, e.g. a living room. Even though the singular form “loudspeaker” or “driver” is commonly used in the following, the described phenomena and concepts in general do also apply to the use of multiple loudspeakers or multiple drivers, even though this is not specifically mentioned everywhere.

Loudspeakers can be optimized during the design and manufacturing process to perform as intended under specific predefined conditions or assumptions (e.g. for a reference position in a reference room, or optimization under anechoic conditions). However, as soon as the loudspeaker is put into a different environment, its performance will be influenced by the environment. This is mainly due to the fact that the sound that is generated by / radiated from the loudspeaker is interacting with and as such is influenced by the surfaces and objects in the loudspeaker’s vicinity. Such influences are e.g. reflection, absorption, diffraction. Especially in the lower frequency range, proximity to boundary surfaces can cause significant changes in the loudspeaker’s performance.

The sound field that actually builds up at a specific listener position is a combination of all contributing sounds, in particular, direct sound from the loudspeaker plus reflected sound from the environment.

Since the interaction between direct sound and reflected sound is specific for individual source-receiver position combinations, the actual performance of the loudspeaker changes both with changing position of the loudspeaker and changing position of the listener within the actual listening environment.

It is such often desired that the loudspeaker is adjusted to the actual listening situation. Such, the performance of a loudspeaker can be adjusted by applying suitable filters for a given loudspeaker position and a given listener position.

Usually the adjustment is done in the state of the art by using a measurement microphone at the listening position and, based on specific test signals, generation of equalization filters.

If sound reproduction in a broader listening area that covers multiple listening positions should be adapted, then usually multiple measurements, e.g., multipoint measurements, are used.

Different averaging approaches considering the multiple measurements can be used to find the best compromise equalization for the whole (measurement) area.

The aforementioned concepts require a user interaction (for initial setup, and they would need it every time the loudspeaker position (for some even if the listener position) is changed). Plus, due to the need to setup a microphone(s) in the listening area, they may be intrusive. Overall, not very user friendly or easy to use. Additionally, for naive users, even that may pose problems, and there is the chance that they do something wrong.

Besides those single-point measurement or multipoint-measurement optimizations, it is possible to mitigate some general influences of listening environments on the loudspeaker performance by rough adjustment concepts that do not require a specific measurement.

E.g. if the loudspeaker is placed close to a wall, this will result in a level increase in the lower frequency range. Some loudspeakers address that by offering dip switches that can activate predefined filters that would tackle such common scenarios.

However, such kinds of settings already require some kind of expert knowledge from the user to choose the correct settings. Furthermore, they are not very flexible.

With the advent of wireless portable loudspeakers that can easily be moved to different positions, concepts for adaption of the loudspeaker to its actual placement that have a beneficial effect in a large listening area are desired. Such an equalization can be achieved by utilizing a scheme that targets a global equalization, which takes into consideration influences of the room on the reproduced/generated soundfield that can be measured in one position but are valid basically all over the room. In the state of the art, methods exist that estimate a global response, which reveals characteristics that pertain throughout the entire listening environment (i.e. they correspond to the average one would get by multiple single point measurements throughout the room). Such, by equalizing those global characteristics, an advantageous adaption of the loudspeaker to the specific room and its specific present setup position can be made which is beneficial for listeners all over the room. These described concepts have been used for automatic adaption of loudspeakers to their environment.

Prior art outlines that the calculation of a global equalization can be based on estimation of sound pressure and velocity to estimate the frequency dependent radiation resistance, in particular, the real part of the frequency dependent radiation impedance.

US 2002/0154785 A1 describes a method and apparatus for controlling the performance of a loudspeaker in a room. The method comprises the steps of determining the acceleration, velocity or displacement of a loudspeaker diaphragm and the sound pressure in front of the diaphragm in a reference acoustic environment, and determining based on these quantities the radiation resistance, radiated acoustic power or real part of the acoustic wave impedance. The same parameters are measured in the actual listening environment, and the ratio of both is used to control a correction filter. The complete procedure is based on the realization that there is a strong link between the way the loudspeaker sounds, in particular in the bass range, and its radiation resistance as a function of frequency, being the real part of the radiation impedance. According to US 2002/0154785 A1, parameters are measured in a first environment and same parameters are measure in a second environment, a ratio of both measurements is taken to define a correction filter. Summarizing, US 2002/0154785 A1 relates to a method for controlling the performance of a loudspeaker in a room wherein in a first acoustic environment the resultant movement of the loudspeaker driver diaphragm and the associated force, arising from the sound field in the room, acting on it are determined by measuring suitable parameters defining a first complex transfer function. In a second acoustic environment a second complex transfer function is determined by measuring the same or different parameters of the loudspeaker driver relating to the room. The ratio between the real parts of the first and second transfer function is used to define the performance of a correction filter. The filter is applied in the signal chain to the loudspeaker driver. WO 00/21331 A1 describes that to make a loudspeaker environmentally adaptive, a measurement of the velocity or acceleration of the loudspeaker diaphragm and the associated sound pressure in front of the diaphragm, an accelerometer and a microphone are needed to determine the radiation resistance of the diaphragm. WO 00/21331 A1 further realized that those two sensors would have to be expensive to ensure consistent behavior over a long lifetime. Such, a way is presented to exchange the accelerometer by another microphone that is placed in small distance from the diaphragm. This is based on the insight that changes in the radiation resistance can be based on a measurement of the sound pressure in two (or more) points spaced differently from the loudspeaker diaphragm. Further, in WO 00/21331 A1 , ways are presented to use only a single microphone which is physically moved to different positions. Summarizing, WO 00/21331 A1 relates to a loudspeaker of the type having sensor means for the determination of the radiation resistance of the diaphragm, expressed by the velocity/acceleration of the loudspeaker diaphragm and the sound pressure in a distance from the diaphragm. Thereby, via a signal processing unit, provide a control signal to a filter unit adjusting the performance of the loudspeaker in an adaptive manner to the acoustical characteristics of the listening room. Said sensors comprise a microphone for detecting said sound pressure. The sensor equipment comprises microphone means for detecting the sound pressure in at least two points differently spaced from the diaphragm, and that carrier means are provided enabling one same microphone to be effectively and successively exposed to the sound pressure in each of the at least two points. In WO 00/21331 A1 , the two measurement points mentioned here really have to be close to the diaphragm. If the distance is getting bigger, the estimation will increasingly fail. Furthermore, WO 00/21331 A1 outlines that it would be sufficient to obtain a reference value i.e. the absolute radiation resistance except for a scaling factor, for comparison with later detections of the sound pressure in the same two (or more) points.

US 2017/0195790 A1 describes a loudspeaker system with an external microphone outside of the loudspeaker’s enclosure, and an internal microphone inside the loudspeaker’s enclosure. A transfer function for an equalization filter is determined responsive to the external and internal microphone. The external microphone(s) [one, two or more] is(are) located to measure acoustic pressure in the vicinity of the driver. The internal microphone is used to indirectly measure volume velocity of the loudspeaker diaphragm.

Summarizing, according to the prior art, the volume velocity is estimated from the gradient of sound pressure in front of the loudspeaker (requires either two very similar measurement devices, or moving parts, or an accelerometer). Global equalization solutions can be based on estimation of the sound pressure in front of the loudspeaker and the volume velocity. The sound pressure can be measured with a microphone close to / in front of the loudspeaker (i.e. in front of the membrane /driver/diaphragm). Volume velocity estimation has been described based on estimating the gradient of sound pressure in front of the loudspeaker (e.g. by using two microphones, or a single microphone with mechanical means to use that single microphone for measurements at two spatially different locations).

The object of the present invention is to provide improved concepts for audio reproduction. The object of the present invention is solved by an apparatus according to claim 1 , by an apparatus according to claim 39, by a method according to claim 44, by a method according to claim 45 and by a computer program according to claim 46.

An apparatus for processing an audio input signal comprising one or more audio input channels to obtain an audio output signal comprising one or more audio output channels according to an embodiment is provided. The apparatus comprises an estimation unit configured to estimate a radiation resistance of each driver of one or more drivers of each loudspeaker of one or more loudspeakers as an estimated radiation resistance; or configured to estimate a radiation impedance of each driver of the one or more drivers of each loudspeaker of the one or more loudspeakers as an estimated radiation impedance, wherein said estimated radiation impedance of said driver comprises estimated information on the radiation resistance of said driver. Moreover, the apparatus comprises a processing unit configured to obtain the one or more audio output channels by processing each audio input channel of the one or more audio input channels depending on the estimated radiation resistance or depending on the estimated radiation impedance of each of the one or more drivers of each of the one or more loudspeakers. To estimate the estimated radiation resistance or the estimated radiation impedance of each driver of the one or more drivers of each loudspeaker of the one or more loudspeakers, the estimation unit is configured to estimate the estimated radiation resistance or the estimated radiation impedance depending on estimated sound pressure information indicating an estimation of sound pressure at said driver of said loudspeaker, and depending on estimated velocity information indicating an estimation of a driver velocity of said driver of said loudspeaker.

Moreover, a method for processing an audio input signal comprising one or more audio input channels to obtain an audio output signal comprising one or more audio output channels according to an embodiment is provided. The method comprises: Estimating a radiation resistance of each driver of one or more drivers of each loudspeaker of one or more loudspeakers as an estimated radiation resistance; or estimating a radiation impedance of each driver of the one or more drivers of each loudspeaker of the one or more loudspeakers as an estimated radiation impedance, wherein said estimated radiation impedance of said driver comprises estimated information on the radiation resistance of said driver. And:

Obtaining the one or more audio output channels by processing each audio input channel of the one or more audio input channels depending on the estimated radiation resistance or depending on the estimated radiation impedance of each of the one or more drivers of each of the one or more loudspeakers.

To estimate the estimated radiation resistance or the estimated radiation impedance of each driver of the one or more drivers of each loudspeaker of the one or more loudspeakers, estimating the estimated radiation resistance or the estimated radiation impedance is conducted depending on estimated sound pressure information indicating an estimation of sound pressure at said driver of said loudspeaker, and depending on estimated velocity information indicating an estimation of a driver velocity of said driver of said loudspeaker.

Furthermore, an apparatus comprising an estimation unit is provided. The estimation unit is configured to estimate a first radiation resistance of each driver of one or more drivers of each loudspeaker of one or more loudspeakers as a first estimated radiation resistance before a first point in time; or is configured to estimate a first radiation impedance of each driver of the one or more drivers of each loudspeaker of the one or more loudspeakers as a first estimated radiation impedance before the first point in time, wherein said first estimated radiation impedance of said driver comprises estimated information on the first radiation resistance of said driver. T o estimate the first estimated radiation resistance or the first estimated radiation impedance of each driver of the one or more drivers of each loudspeaker of the one or more loudspeakers, the estimation unit is configured to estimate the first estimated radiation resistance or the first estimated radiation impedance depending on first estimated sound pressure information indicating an estimation of sound pressure at said driver of said loudspeaker before the first point in time, and depending on first estimated velocity information indicating an estimation of a first driver velocity of said driver of said loudspeaker before the first point in time. Moreover, the estimation unit is configured to estimate a second radiation resistance of each driver of the one or more drivers of each loudspeaker of the one or more loudspeakers as a second estimated radiation resistance after a second point in time; or is configured to estimate a second radiation impedance of each driver of the one or more drivers of each loudspeaker of the one or more loudspeakers as a second estimated radiation impedance after the second point in time, wherein said second estimated radiation impedance of said driver comprises estimated information on the second radiation resistance of said driver. The second point in time occurs after the first point in time. To estimate the second estimated radiation resistance or the second estimated radiation impedance of each driver of the one or more drivers of each loudspeaker of the one or more loudspeakers, the estimation unit is configured to estimate the second estimated radiation resistance or the second estimated radiation impedance depending on second estimated sound pressure information indicating an estimation of sound pressure at said driver of said loudspeaker after the second point in time, and depending on second estimated velocity information indicating an estimation of a second driver velocity of said driver of said loudspeaker after the second point in time. Furthermore, the estimation unit is configured to determine and to output whether the apparatus is in a first state or whether the apparatus is in a second state depending on a radiation resistance difference indicating a difference between the second estimated radiation resistance and the first estimated radiation resistance, or depending on a radiation impedance difference indicating a difference between the second estimated radiation impedance and the first estimated radiation impedance. The second state indicates that the apparatus is malfunctioning or that the apparatus has been relocated. The first state indicates that the apparatus is functioning and that the apparatus has not been relocated.

Moreover, a method is provided. The method comprises:

Estimating a first radiation resistance of each driver of one or more drivers of each loudspeaker of one or more loudspeakers as a first estimated radiation resistance before a first point in time; or estimating a first radiation impedance of each driver of the one or more drivers of each loudspeaker of the one or more loudspeakers as a first estimated radiation impedance before the first point in time, wherein said first estimated radiation impedance of said driver comprises estimated information on the first radiation resistance of said driver; wherein to estimate the first estimated radiation resistance or the first estimated radiation impedance of each driver of the one or more drivers of each loudspeaker of the one or more loudspeakers. Estimating the first estimated radiation resistance or the first estimated radiation impedance is conducted depending on first estimated sound pressure information indicating an estimation of sound pressure at said driver of said loudspeaker before the first point in time, and depending on first estimated velocity information indicating an estimation of a first driver velocity of said driver of said loudspeaker before the first point in time.

Estimating a second radiation resistance of each driver of the one or more drivers of each loudspeaker of the one or more loudspeakers as a second estimated radiation resistance after a second point in time; or estimating a second radiation impedance of each driver of the one or more drivers of each loudspeaker of the one or more loudspeakers as a second estimated radiation impedance after the second point in time, wherein said second estimated radiation impedance of said driver comprises estimated information on the second radiation resistance of said driver; wherein to estimate the second estimated radiation resistance or the second estimated radiation impedance of each driver of the one or more drivers of each loudspeaker of the one or more loudspeakers. The second point in time occurs after the first point in time. Estimating the second estimated radiation resistance or the second estimated radiation impedance is conducted depending on second estimated sound pressure information indicating an estimation of sound pressure at said driver of said loudspeaker after the second point in time, and depending on second estimated velocity information indicating an estimation of a second driver velocity of said driver of said loudspeaker after the second point in time. And;

Determining and outputting whether the apparatus is in a first state or whether the apparatus is in a second state depending on a radiation resistance difference indicating a difference between the second estimated radiation resistance and the first estimated radiation resistance, or depending on a radiation impedance difference indicating a difference between the second estimated radiation impedance and the first estimated radiation impedance, wherein the second state indicates that the apparatus is malfunctioning or that the apparatus has been relocated, and wherein the first state indicates that the apparatus is functioning and that the apparatus has not been relocated.

Furthermore, a computer program is provided, which is configured to implement one of the above-described methods when being executed on a computer or signal processor. In the following, embodiments of the present invention are described in more detail with reference to the figures, in which:

Fig. 1 illustrates an apparatus according to an embodiment.

Fig. 2 illustrates a system according to an embodiment.

Fig. 3 illustrates a loudspeaker of an example with an indication of three different measurement positions.

Fig. 4 depicts a high-level illustration of an embodiment.

Fig. 5 illustrates some example real world results for a specific loudspeaker in different positions in the same room according to embodiments.

Fig. 6 illustrates the magnitude-response of the global equalization filter after interpolation according to a specific example, and further illustrates band limiting for a specific example.

Fig. 7 depicts a high-resolution display of an unprocessed filter prototype according to an embodiment.

Fig. 8 illustrates a usage of models to estimate the parameters according to an embodiment.

Fig. 9 illustrates a linear lumped parameter model according to an embodiment.

Fig. 10 illustrates a side view of an alternative loudspeaker layout with drivers/transducers at four sides according to an embodiment.

Fig. 11 illustrates a top view of an alternative loudspeaker layout with drivers/transducers at four sides according to an embodiment.

Fig. 12 illustrates an alternative loudspeaker layout being a soundbar-type with multiple microphones according to an embodiment. Fig. 13 illustrates an example of a loudspeaker positioned on a surface according to an embodiment.

Fig. 14 illustrates a top view of a loudspeaker showing potential positions for single or multiple microphones according to an embodiment.

Fig. 15 illustrates a side view of a loudspeaker showing potential positions for single or multiple microphones according to an embodiment.

Fig. 16 illustrates another side view of a loudspeaker showing potential positions for single or multiple microphones according to another embodiment.

Fig. 1 illustrates an apparatus 100 for processing an audio input signal comprising one or more audio input channels to obtain an audio output signal comprising one or more audio output channels according to an embodiment.

The apparatus 100 comprises an estimation unit 110. The estimation unit 110 is configured to estimate a radiation resistance of each driver of one or more drivers of each loudspeaker of one or more loudspeakers as an estimated radiation resistance; or is configured to estimate a radiation impedance of each driver of the one or more drivers of each loudspeaker of the one or more loudspeakers as an estimated radiation impedance. Said estimated radiation impedance of said driver comprises estimated information on the radiation resistance of said driver.

Moreover, the apparatus 100 comprises a processing unit 120 configured to obtain the one or more audio output channels by processing each audio input channel of the one or more audio input channels depending on the estimated radiation resistance or depending on the estimated radiation impedance of each of the one or more drivers of each of the one or more loudspeakers.

To estimate the estimated radiation resistance or the estimated radiation impedance of each driver of the one or more drivers of each loudspeaker of the one or more loudspeakers, the estimation unit 110 is configured to estimate the estimated radiation resistance or the estimated radiation impedance depending on estimated sound pressure information indicating an estimation of sound pressure at said driver of said loudspeaker, and depending on estimated velocity information indicating an estimation of a driver velocity of said driver of said loudspeaker.

For example, a radiation impedance of a driver may, e.g., be represented in a complex domain, e.g., by a plurality of complex values (e.g., elements of ) . A radiation resistance of a driver may, e.g., be represented in a real domain, e.g., by a plurality of real values (e.g., elements of ). For example, for each complex value of a plurality of complex values of the radiation impedance of a driver, the real part (in contrast to the imaginary part) of said complex value may, e.g., represent the information on the radiation resistance that is provided by said complex value. Or, in other words, if a plurality of complex values represent the information on the radiation impedance, the real parts of the plurality of complex values may, e.g., represent the information on the radiation resistance.

In some of the embodiments, each of the one or more audio input channels and the one or more audio output signals may, e.g., by one or more (traditional/ordinary) audio channel signals.

In some other embodiments, each of the one or more audio input channels and the one or more audio output signals may, e.g., by one or more audio object signals.

In some further embodiments, the one or more audio input channels and the one or more audio output channels may, e.g., comprise at least one traditional/ordinary audio channel signal and at least one audio object signal.

The one or more audio object signals and/or the at least one audio object signal mentioned before may, for example, be one or more Spatial Audio Object Coding (SAOC) object signals.

In some other embodiments, at least one of the one or more audio input channels and the one or more audio output signals may, e.g., comprise scene based audio information.

In some embodiments, a loudspeaker may, e.g., comprise a transducer to convert electric signals into sound. Such a transducer (of a specific building-type) may, e.g., comprise a cone/diaphragm. Such a transducer may, e.g., be built into an enclosure. Thus, according to some embodiments, a loudspeaker may, e.g., comprise a transducer and an enclosure.

In some embodiments, a driver may, e.g., be implemented as a moving diaphragm of a transducer.

According to some embodiments, the one or more loudspeakers mentioned here and/or the one or more microphones mentioned here may, e.g., be installed in a soundbar, in a smart speaker, in a TV, in a laptop, in a single loudspeaker system.

In some embodiments at least one of the one or more loudspeakers may, e.g., be a subwoofer.

According to an embodiment, the one or more microphones may, e.g., be spaced apart from said loudspeaker or spaced apart from said driver of said loudspeaker.

In an embodiment, to estimate the estimated radiation resistance or the estimated radiation impedance of said driver of said loudspeaker, the estimation unit 110 may, e.g., be configured to estimate the estimated radiationx resistance or the estimated radiation impedance by estimating estimated sound pressure information indicating an estimation of sound pressure at said driver of said loudspeaker, and/or by estimating estimated velocity information indicating an estimation of a driver velocity of said driver of said loudspeaker.

In an embodiment, the estimation unit 110 may, e.g., be configured to estimate the estimated sound pressure information such that the estimated sound pressure information is represented in a spectral domain; and/or the estimation unit 110 may, e.g., be configured to estimate the estimated velocity information such that the estimated velocity information is represented in the spectral domain. Moreover, the estimation unit 110 may, e.g., be configured to estimate the estimated radiation resistance or the estimated radiation impedance of said driver of said loudspeaker such that the estimated radiation resistance or the estimated radiation impedance of said driver of said loudspeaker is represented in the spectral domain.

In an embodiment, the estimation unit 110 may, e.g., be configured to estimate the estimated sound pressure information depending on a sound pressure P _m3 at a microphone of the one or more microphones. According to an embodiment, the estimation unit 110 may, e.g., be configured to estimate the estimated velocity information depending on a current through a loudspeaker driver coil of said driver of said loudspeaker.

In an embodiment, the estimation unit 110 may, e.g., be configured to estimate the estimated velocity information depending on an electrical resistance R _e , a coil inductance L _e , a force factor Bl, a mechanical mass M, a total stiffness K, a mechanical resistance R _m . v indicates the cone velocity / driver velocity .

According to an embodiment, the estimation unit 110 may, e.g., be configured to determine the estimated velocity information depending on an equation system, being defined according to: wherein u(t) indicates an excitation signal, wherein t indicates time, wherein x indicates an axial displacement of the loudspeaker diaphragm of said loudspeaker, wherein I indicates the current through the loudspeaker driver coil of said driver of said loudspeaker, wherein the notation represents the first-order derivative with respect to time.

In an embodiment, the estimation unit 110 may, e.g., be configured to solve the equation system using a fourth-order Runge-Kutta method.

According to another embodiment, the estimated velocity information may, e.g., be stored within the apparatus 100. In an embodiment, the estimated velocity information may, e.g., be stored in a look-up table which is stored within the apparatus 100. The estimation unit 110 may, e.g., be configured to derive the estimated velocity information from the look-up table.

According to an embodiment, the estimation unit 110 may, e.g., be configured to determine linear parameters of said driver of said loudspeaker by solving a minimization problem / an optimization problem to estimate the estimated radiation resistance or the estimated radiation impedance of said driver of said loudspeaker. E.g., the linear parameters may, e.g., be used for modelling as described herein.

In an embodiment, the estimation unit 110 may, e.g., be configured to use said estimated sound pressure information to estimate said estimated velocity information.

According to an embodiment, the estimation unit 110 may, e.g., be configured to employ wherein v is a time derivative of the estimated velocity information, wherein V is a gradient operator, wherein p is the estimated sound pressure information in the time domain, wherein p is a medium density.

For example, p may, e.g., indicate the pressure information in the time domain; whereas P may, e.g., indicate the pressure information in the spectral domain, e.g., frequency domain.

In an embodiment, the processing unit 120 may, e.g., be configured to determine a difference between the estimated radiation resistance of said driver of said loudspeaker and a predefined radiation resistance. The processing unit 120 may, e.g., be configured to process the one or more audio input channels depending on the difference between the estimated radiation resistance of said driver of said loudspeaker and the predefined radiation resistance.

According to an embodiment, the processing unit 120 may, e.g., be configured to modify a spectral shape of at least one of the one or more audio input channels depending on the difference between the estimated radiation resistance of said driver of said loudspeaker and the predefined radiation resistance to obtain the one or more audio output signals. In an embodiment, the processing unit 120 may, e.g., be configured to determine a spectral modification factor for each spectral band of a plurality of spectral bands depending on the difference between the estimated radiation resistance of said driver of said loudspeaker and the predefined radiation resistance for said spectral band. For each audio input channel of the one or more audio input channels, to obtain one of the one or more audio output channels, the processing unit 120 may, e.g., be configured to apply the spectral modification factor of each spectral band of the plurality of spectral bands, on said spectral band of said audio input channel.

According to an embodiment, the processing unit 120 may, e.g., be configured to determine the difference between the estimated radiation resistance of said driver of said loudspeaker and the predefined radiation resistance according to wherein H _raw (ω) indicates said difference, wherein R _r (ω) indicates the estimated radiation resistance, wherein indicates the predefined radiation resistance, wherein w indicates an angular frequency.

In an embodiment, the processing unit 120 may, e.g., be configured to apply a smoothing operation on said difference being an unprocessed filter prototype to obtain a smoothed filter prototype. Moreover, the processing unit 120 may, e.g., be configured to apply the smoothed filter prototype on at least one of the one or more audio input channels to obtain at least one of the one or more audio output channels.

According to an embodiment, the processing unit 120 may, e.g., be configured to apply a global equalizer on at least one of one or more audio input signal to obtain at least one intermediate signal. Moreover, the processing unit 120 may, e.g., be configured to determine a relative sound power in a spectral domain from the estimated radiation resistance or from the estimated radiation impedance. Furthermore, the processing unit 120 may, e.g., be configured to determine one or more peaks (e.g., one or more local maxima) within the relative sound power in the spectral domain. Moreover, the processing unit 120 may, e.g., be configured to apply a further equalizer on the at least one intermediate signal depending on the one or more peaks within the relative sound power in the spectral domain to obtain at least one of the one or more audio output channels.

In an embodiment, the estimation unit 110 may, e.g., be configured to estimate the estimated sound pressure information depending on captured sound pressure information recorded by one or more microphones.

According to an embodiment, the one or more microphones are two or more microphones. The estimation unit 110 may, e.g., be configured to receive the captured sound pressure information from the two or more microphones. Moreover, the estimation unit 110 may, e.g., be configured to use the captured sound pressure information from only one of the two or more microphones to determine the estimated sound pressure information. Furthermore, the estimation unit 110 may, e.g., be configured to not use the captured sound pressure information from the other microphones of the two or more microphones to determine the estimated sound pressure information.

In an embodiment, the one or more microphones are two or more microphones. The estimation unit 110 may, e.g., be configured to receive the captured sound pressure information from the two or more microphones. Moreover, the estimation unit 110 may, e.g., be configured to determine an average or a weighted average of the captured sound pressure information from the two or more microphones, and to determine the estimated sound pressure information using the average or the weighted average of the captured sound pressure information.

For example, if there are two sound pressure values p ₁ and p ₂ , the average may, e.g., be: a = 0.5 p ₁ + 0.5 P ₂ ; and the weighted average a _w with weights w ₁ and w ₂ may, e.g., be a _w = w ₁ p ₁ + W ₂ P ₂ . For example 0 < w ₁ < 1 and w ₂ = 1 - W ₁ .

For example, if there are three sound pressure values pi and p _å and p ₃ , the average may, e.g., be: a = 1/3 p ₁ + 1/3 p ₂ + 1/3 p ₃ ; and the weighted average a _w with weights Wi and w ₂ and W ₃ may, e.g., be a _w = w ₁ p ₁ + w ₂ p ₂ + w ₃ p ₃ . For example 0 < w ₁ < 1; 0 < w ₂ < 1;

0 < W ₁ + w ₂ < 1 and w ₃ = 1 - w ₁ - w ₂ .

According to an embodiment, the one or more microphones may, e.g., be two or more microphones. The one or more loudspeakers may, e.g., be two or more loudspeakers and/or at least one of the one or more loudspeakers may, e.g., comprise two or more drivers. The estimation unit 110 may, e.g., be configured to receive the captured sound pressure information from the two or more microphones. Moreover, the estimation unit 110 may, e.g., be configured to determine, for each driver of the one or more drivers of each loudspeaker of the one or more loudspeakers, a weighted average of the captured sound pressure information from the two or more microphones, and to determine the estimated sound pressure information using the weighted average of the captured sound pressure information, wherein the estimation unit 110 may, e.g., be configured to determine said weighted average depending on a plurality of weights, wherein each weight of the plurality of weights depends on a position of said driver and depends on a position of each of the two or more microphones.

According to an embodiment, the one or more microphones may, e.g., be two or more microphones. The one or more loudspeakers may, e.g., be two or more loudspeakers and/or at least one of the one or more loudspeakers may, e.g., comprise two or more drivers. For each driver of the one or more drivers of the one or more loudspeakers, the estimation unit 110 may, e.g., be configured to select one of the two or more microphones as a selected microphone For said driver, the estimation unit 110 may, e.g., be configured to use the captured sound pressure information from the selected microphone to determine the estimated sound pressure information. Moreover, for said driver, the estimation unit 110 may, e.g., be configured to not use the captured sound pressure information from the other microphones of the two or more microphones to determine the estimated sound pressure information.

In an embodiment, the estimation unit 110 may, e.g., be configured to determine the estimated sound pressure information using a complex transfer function.

According to an embodiment, the estimation unit 110 may, e.g., be configured to determine the estimated sound pressure information depending on P ≈ P _m3 /H , wherein P indicates the estimated sound pressure information, wherein P _m3 indicates the captured sound pressure information, wherein H indicates the complex transfer function being defined as wherein w indicates an angular frequency, (for example, ω ∈ ), wherein P _src indicates an imposed sound pressure at said loudspeaker, wherein P _rec indicates an estimated/simulated sound pressure at said one of the one or more microphones that is present when the sound pressure P _src exists at the loudspeaker. P _src and P _rec may, e.g., be obtained from an acoustic model.

In an embodiment, for each driver of the one or more drivers of the one or more loudspeakers, the estimation unit 110 may, e.g., be configured to select one of the two or more microphones as a selected microphone depending on a position of said driver and depending on a position of each of the two or more microphones.

According to an embodiment, the one or more audio input channels may, e.g., be two or more audio input channels, and the one or more audio output channels may, e.g., be two or more audio output channels. The processing unit 120 may, e.g., be configured to obtain at least two of the two or more audio output channels by determining, depending on the estimated radiation resistance or depending on the estimated radiation impedance of at least one of the one or more drivers of each of the one or more loudspeakers, individual modification information for each audio input channel of the at least two of the two or more audio input channels; and by applying the individual modification information for each audio input channel of the at least two of the two or more audio input channels on said audio input channel.

Thus, in such an embodiment, different audio input channels are treated differently. For example, it may be desirable for a 5.1 audio input signal to enhance bass frequencies for the LFE channel, and to reduce bass in other channels.

Such, if the estimated radiation resistance indicates e.g. that the positioning of the loudspeaker results in a boost of bass frequencies, this could e.g. beneficially be preserved for an LFE or subwoofer channel, while it would be reduced for the other channels.

Moreover, it is not always desirable to suppress room acoustic properties. Some audio input channels may, e.g., be modified such that room acoustic properties are beneficially be taken into account.

For example, sometimes, it may be useful to enhance or boost high-frequency audio components, e.g., that are reproduced using one or more tweeters, instead of reducing low- frequency/bass audio components, as such a strategy may, e.g., result in a more impressive sound experience, or e.g. because the loudspeaker can such produce an overall higher level / gain while the defined adaption of the frequency curve still follows a defined target.

Moreover, different drivers of a loudspeaker can be intended/optimized for different frequency ranges, for example, woofers, full-range drivers, tweeters, etc.

This differentiation can be taken into account in the design of the one or more reference curves / target curves / defined targets.

According to an embodiment, at least one of the one or more microphones 300 is not located on a main radiation direction of any of the one or more loudspeakers 200.

In an embodiment, at least one of the one or more microphones 300 has not a direct line of sight to any of the one or more loudspeakers 200.

According to an embodiment, for each microphone of the one or more microphones, a predefined distance between said microphone and the loudspeaker may, e.g., be at least 10 centimetres, e.g., at least 20 centimetres, e.g., at least 50 centimetres, e.g., at least 1 meter. Even with these distances, the concepts of the invention still work, e.g., due to the provided estimation concepts.

According to an embodiment, the estimation unit 110 may, e.g., be configured to update the estimated radiation resistance or the estimated radiation impedance of the one or more drivers of the one or more loudspeakers at/during initialization and when requested and at/during runtime.

For example, the estimated radiation resistance or the estimated radiation impedance may, e.g., be estimated, when the apparatus is moved in a listening environment, e.g., in a room, and may, e.g., also be periodically updated (and not only at initialization).

In an embodiment, the estimated radiation resistance is a first estimated radiation resistance before a first point in time, or the estimated radiation impedance is a first estimated radiation impedance before the first point in time. The estimation unit 110 may, e.g., be configured to estimate a second radiation resistance of each driver of the one or more drivers of each loudspeaker of the one or more loudspeakers as a second estimated radiation resistance after a second point in time; or is configured to estimate a second radiation impedance of each driver of the one or more drivers of each loudspeaker of the one or more loudspeakers as a second estimated radiation impedance after the second point in time, wherein said second estimated radiation impedance of said driver comprises estimated information on the second radiation resistance of said driver. The second point in time occurs after the first point in time. To estimate the second estimated radiation resistance or the second estimated radiation impedance of each driver of the one or more drivers of each loudspeaker of the one or more loudspeakers, the estimation unit 110 may, e.g., be configured to estimate the second estimated radiation resistance or the second estimated radiation impedance depending on second estimated sound pressure information indicating an estimation of a second sound pressure at said driver of said loudspeaker, and depending on second estimated velocity information indicating an estimation of a second driver velocity of said driver of said loudspeaker. Moreover, the estimation unit 110 may, e.g., be configured to determine and to output whether the apparatus 100 is in a first state or whether the apparatus 100 is in a second state depending on a radiation resistance difference indicating a difference between the second estimated radiation resistance and the first estimated radiation resistance, or depending on a radiation impedance difference indicating a difference between the second estimated radiation impedance and the first estimated radiation impedance. The second state indicates that the apparatus 100 is malfunctioning or that the apparatus 100 has been relocated. The first state indicates that the apparatus 100 is functioning and that the apparatus 100 has not been relocated.

According to an embodiment, the estimation unit 110 may, e.g., be configured to estimate the second estimated sound pressure information depending on captured second sound pressure information recorded by the one or more microphones; and/or the estimation unit 110 may, e.g., be configured to estimate the second estimated velocity information depending on a second current through the loudspeaker driver coil of said driver of said loudspeaker after the second point in time.

In an embodiment, additionally other means, for example, from one or more gyroscopes, or other information that has been gathered from the pressure measurement, may, e.g., also be used as an indication that the device has been moved.

Fig. 2 illustrates a system according to an embodiment. The system comprises the apparatus 100 as described above with respect to Fig. 1 and the loudspeaker 200 referred to above. The loudspeaker 200 is configured to output at least one of the one or more audio output channels.

In an embodiment, the system may, e.g., further comprise the one or more microphones 300 referred to above.

In the following, further concepts and further embodiments of the present invention are provided.

According to some of the embodiments, the microphone does not have to be positioned close to or in front of the loudspeaker diaphragm to measure the sound pressure.

In some of the embodiments, it may, e.g., be assumed that at least one microphone is present somewhere on the enclosure of the loudspeaker. The at least one microphone may, e.g., also be close by the loudspeaker, as long as the setup is known, so that the sound transmission (path) can be simulated from the diaphragm to the at least one microphone. By including insight from simulations of that specific arrangement, the sound pressure that exists close to the diaphragm can be inferred.

Some of the embodiments may, e.g., not need sound pressure gradient measurements (requiring two microphones) or accelerometer measurements to measure the volume velocity.

In some of the embodiments, the volume velocity may, e.g., be estimated based on an electro-mechanical model of the loudspeaker. This model is fed with the output of a voltage/current measurement that is gained at the loudspeaker ports during operation.

Some of the embodiments provide concepts that can automatically adapt the playback performance of an audio reproduction system to a playback environment. This automatic adaption of the playback system may, e.g., happen in form of an, e.g., automatic, calibration of the timbral characteristics of the playback system to be best suited for the current listening environment and loudspeaker position.

Usually, during the design, manufacturing, tuning of a new device, the geometry of the enclosure and the arrangement of the transducers (sources and receivers, for example, (drivers of) loudspeakers and/or microphones) are known. Some of the embodiments may, e.g., use these known properties to achieve a beneficial method of calibrating a sound system in an environment.

According to some of the embodiments, estimation (via simulation) of acoustic quantities that are required to compute the radiation impedance of a loudspeaker in a room may, e.g., be conducted. In contrast, previous methods relied on measurement of the needed parameters.

In some of the embodiments, a concept is provided to estimate the radiation resistance, or rather the sound pressure and velocity, which has advantages compared to the state of the art, when used for specific classes of reproduction devices.

Some of the embodiments use one or more modeling approaches, and the necessity of using a specific microphone to measure the sound pressure close to the membrane, as well as the necessity of using two microphones or other sophisticated tools or setups to measure the velocity are made obsolete.

In some of the embodiments, the microphones may, e.g., not be directly in front of the diaphragm. For example, the microphones may, e.g., be farther away than a few centimeters from the diaphragm.

In contrast to the prior art, some of the embodiments only need a sound pressure estimate in one point.

Some of the embodiments may, e.g., not need an accelerometer, and some of the embodiments may, e.g., not need to move the microphone and may, e.g., not have to be close to the diaphragm.

In the following, details and ideas of particular embodiments of the present invention are described.

At first, details of radiation impedance calculation and radiation resistance calculation are provided.

The radiation impedance Z(w) is given by the ratio of the sound pressure at the driver R(w) to the normal velocity of the driver V(ω), as follows: wherein C is a constant related to the area of the driver diaphragm.

Fig. 3 illustrates a loudspeaker of an example with an indication of three different (sound pressure) measurement positions. Inter alia, Fig. 3 shows a two point measurement by m _i and m ₂ , where m ₁ and m ₂ positioned closely in front of the speaker diaphragm correspond to the two microphones / the two measurement positions.

In other embodiments not depicted by Fig. 3, two or more microphones are used, where one microphone is positioned inside the loudspeaker enclosure. An accelerometer is placed on the loudspeaker diaphragm.

Returning to Fig. 3, the sound pressure at the driver surface is given (approximately), as indicated in Fig. 3, by the sound pressure P _m1 measured at position m ₁ or by the sound pressure P _m2 measured at position m ₂ , or by an average of P _m1 and P _m2 . An approximate normal velocity can be computed from the sound pressure P _m1 measured at position m ₁ and the sound pressure P _m2 measured at position m ₂ , using formula wherein ω is the angular frequency, p is the medium density, i is the imaginary unit, and x is the axial distance from the center of the driver diaphragm (in particular, x _m1 is the axial distance at position m ₁ from the center of the driver diaphragm; x _m2 is the axial distance at position m ₂ from the center of the driver diaphragm).

The radiation impedance Z is calculated using Thus, the acoustic quantities that may, e.g., to be estimated to compute the (acoustic) radiation impedance of a loudspeaker in a closed room are, e.g., the loudspeaker driver’s axial velocity, V, and the acoustic/sound pressure, P, at the driver’s surface.

In some of the embodiments, the current through the loudspeaker driver coil, and the acoustic/sound pressure at a single point external to the loudspeaker enclosure, are measured and used as input data for the estimation of V and P. Here, ..external to the loudspeaker enclosure” may, e.g., refer to a microphone that is preferably positioned at a known and fixed position at or very close to the loudspeakers enclosure, so that the known properties of the transducer and position can be included in the simulation.

The driver velocity and the sound pressure are not directly measured close to the driver. Instead, those values are estimated/approximated. To estimate the velocity, a (lumped) electro-mechanical parameter model is used.

To estimate the pressure, an acoustic model is used.

The acoustic models can e.g. be wave base methods like FEM (Finite Element Method), FDM (Finite Difference Method), BEM (Boundary Element Method), or in the most simple case only a (crude) spherical wave model assumption.

The sound pressure may, e.g., be modeled based on the distance (e.g., radius r) from the diaphragm, e.g., based on where k is the wave number, and Q(ω) is the source signal; or based on where k is the wave number, Q(ω) is the source signal, and a is a term that takes into account e.g. geometrical spreading, directivity of the drivers, room acoustics that have an influence on the damping behavior. For example, a ∈ M In other words, in some of the embodiments, at least one of the measured current through the loudspeaker driver coil and the acoustic/sound pressure at a single point may, e.g., be used as input data for an electro-mechanical model and/or an acoustic model respectively, to gain approximations / estimates of V, and/or P, respectively.

Some models or methods that are used to estimate the estimated velocity may introduce errors that have an effect on the estimated phase of the estimated velocity. To avoid the introduction of such errors, possible solutions include choosing more detailed models, or more accurate numerical methods.

However, in an embodiment, this problem may, e.g., be advantageously be avoided by assuming that the phases of the particle velocity and the acoustic pressure at the driver are related, for example, according to the continuity of momentum: where p is the medium density.

According to an embodiment, the phase of the velocity may, e.g., be estimated from the phase of the estimated pressure.

In a particular embodiment, in addition to what has been described before, the estimated pressure may, e.g., be used to further refine the estimated velocity, for example, such that, the estimation of the velocity does not only depend on the measured current, but may, e.g., additionally depend on information gained from the estimated pressure. This yields refined estimates of the estimated impedance resistance.

Fig. 4 depicts a high-level illustration of an embodiment.

The block RS represents a device to measure the current out of the amplifier / flowing through the driver coil.

This can be achieved by measuring the voltage drop across a resistor, e.g. a shunt resistor. If switch 410 is switched on, the current, measured by the block RS, is fed into an estimation unit to estimate the radiation impedance or the radiation resistance. If the switch 410 is switched off, the measured current is not fed into the estimation unit, and no estimation of the radiation impedance or the radiation resistance takes place.

TF is the transfer path / transfer function from the diaphragm S1 to the microphone m3 (see Fig. 3), which is simulated to gain an estimate of the sound pressure in front of S1.

In the estimation unit, the measured current and the measured sound pressure are fed to the electro-mechanical model and the acoustical model to give estimates of V and P, respectively. Based on those, the radiation impedance or the radiation resistance is calculated to perform global equalization based on a comparison to a theoretical reference curve or a pre-defined (reference) curve.

Fig. 5 illustrates some example real world results of estimated radiation resistances for a specific loudspeaker in different positions in the same room, in relation to the theoretical radiation resistance (predefined radiation resistance) according to embodiments.

Instead of the theoretical radiation impedance curve, any other reference curve may, e.g., be defined, based on which the desired equalizer (EQ) settings may, e.g., be calculated.

The EQ that may, e.g., be used to compensate for the room effects may, e.g., then be based on a comparison of the estimated radiation impedance to, for example, the theoretical radiation impedance; or based on a comparison of the estimated radiation resistance to, e.g., the theoretical radiation resistance.

In some of the embodiments, smoothed versions of the estimated radiation resistance may, e.g., be used to calculate compensation filter curves. In a particular embodiment, a reference radiation resistance curve may, e.g., be selected to perform global equalization by comparing the estimated radiation resistance to a target curve, which may be either pre-defined or a theoretical one. For instance, a free- field radiation resistance formula may be used for this purpose, which may, for example, be defined as: where 5 is the diaphragm area of the loudspeaker and c is the speed of sound.

Fig. 6 shows a real-world example of a radiation resistance estimation in comparison to the free-field reference curve, and the calculated global equalization filter, for a loudspeaker which has been positioned at the corner of a room.

The initial unprocessed filter prototype H _raw (ω) for global equalization may, for example, be computed according to:

For example, a smoothed version H _smooth (ω) of this filter curve H _raw (ω) may, e.g., be used to calculate the final compensation filter, which may, for example, be obtained by smoothing methods, for example, by using octave-band smoothing. The smoothed version of the filter for the specific example is also shown in Fig. 6, where a 1 -octave-band smoothing was applied.

In an embodiment, the frequency resolution may, e.g., be chosen, and may, e.g., be kept unchanged throughout the EQ (equalizer) filter computation.

In another embodiment, to match a pre-defined number of FIR filter taps, interpolation may, e.g., be applied to the smoothed filter, resulting in a coarser frequency resolution.

According to an embodiment, a frequency limiter may, for example, also be applied to restrict the equalization into a specified frequency range. Frequency limiting may, according to an embodiment, for example, be implemented by applying a bandpass filter to the magnitude-response of the EQ filter.

Flere, Fig. 6 illustrates the magnitude-response of the global equalization filter after interpolation (number of filter taps: N = 4096) according to a specific example, and further illustrates band limiting (40 Hz ↔ 500 Hz) H _EQ (ω) that may, e.g., be applied in the specific example. The phase-response of the FIR filter H _EQ (ω) may, for example, be obtained through the computation of the cepstrum to realize a minimum-phase version. The FIR filter taps h _EQ (n) may, for example, be computed by taking the inverse fast Fourier transform (IFFT), for example, according to: h _EQ (n) = IFFT{H _eq (ω)}. (9)

In a further embodiment, the EQ generation may be conducted in another way compared to the EQ generation described above. Such a further embodiment is particular advantageous, if the radiation impedance estimation in a specific room reveals specific problematic frequencies in the low frequency region that stick out, which are often called dominant modes. Such dominant modes can appear if unfavorable combinations of room dimension are present, that boost specific frequencies excessively strong, and/or if the loudspeaker is placed in a position where it excites specific room modes.

Since such excitation of specific room modes leads to audible ringing / resonance / excessively long decay of specific frequency regions that may influence the listening experience unfavorably, it is advantageous to specifically take care of mitigating those modal effects.

As an example, Fig. 7 depicts a high-resolution display of an unprocessed filter prototype according to an embodiment. To better reveal the specific modal issue, in this case, the inverse of the initial unprocessed filter prototype, for example, defined as: indicates the excessive relative sound power in comparison to the reference curve, which is displayed in dB scale.

In the plot of Fig. 7, the described modal behavior can clearly be identified in the region around 57 Hz (indicated by the red circle). To tackle such modal behavior, usually high-Q filters are necessary.

One example of how such a modal behavior equalization could be performed is, e.g., to apply a smoother global EQ as described before in a first stage, and then apply a specific high-Q modal EQ to equalize the specific peaks that were identified in the high frequency resolution analyses.

In another embodiment, the above mentioned modal EQ can be applied using as single loudspeaker to compensate for modal effects.

Multiple loudspeakers can be used to compensate low frequency modal effects in rooms.

A first loudspeaker and at least one additional loudspeaker(s) are positioned in a room, and the modal behavior is controlled by sound fed into the at least one additional loudspeaker(s).

With the method of radiation impedance estimation described herein, such a method using multiple loudspeakers can be beneficially applied, since the necessary identification of the problematic frequency ranges to be equalized can be performed, suitable additional loudspeakers that would be applicable to compensate the detected problematic frequency range(s) can be automatically identified and selected, and a continuous control of the effect of the application of the compensation method can be performed.

Some of the embodiments are implemented such that they are capable of conducting at least one of the above described methods for equalizer generation / equalizer determination.

Further embodiments are implemented such that they are capable of conducting more than one of the above described methods for equalizer generation / equalizer determination, and select one of the methods for equalizer generation / equalizer determination. For example, that one of the methods for equalizer generation / equalizer determination may, e.g., be selected depending on an environment, where the apparatus is used. E.g., one of the methods for equalizer generation / equalizer determination is selected that is most suitable for a particular environment, where the apparatus is used.

Fig. 8 illustrates a usage of models to estimate the parameters according to an embodiment.

In the following, estimating the driver velocity according to some of the embodiments is described. Once the current has been measured, using, for example, the voltage drop across a shunt resistor, a model description of the loudspeaker is used to estimate the normal velocity of the driver.

In an embodiment, the velocity may, e.g., be determined by searching for model parameters that minimize the error between the measured and simulated currents.

Different model descriptions of loudspeakers exist. In the following, the estimation process is described based on one exemplifying, specific model. Actually, this model may, for example, be only valid at low frequencies, but for the given application this is sufficient, since, in particular embodiments, only the low frequency behavior may, e.g., be intended to be equalized. In other embodiments, other models may, e.g., similarly be used.

The electro-mechanical (e.g., linear, e.g., lumped) parameter model of a loudspeaker driver, used as an example here, is shown in Fig. 9.

Fig. 9 illustrates a (e.g., linear, e.g., lumped) parameter model according to an embodiment.

The elements on the electrical side (left part of the sketch Fig. 9) are the driving voltage u(t), the electrical resistance R _e , the coil inductance _e , and the product of the force factor Bl and the cone velocity v (t).

On the mechanical side (right part of the sketch in Fig. 9), the elements are the product of Bl and the current /, the mechanical mass M, the total stiffness K, and the mechanical resistance R _m .

The following two coupled equations describe the model mathematically: and

BlI = Ma + R _m v + Kx , (12) in which the acceleration is given by

Equations (11) and (12) can be written in State Space representation as: where the notation represents the first-order derivative with respect to time x indicates an axial displacement of the loudspeaker diaphragm of said loudspeaker.

The equation system (14) may, e.g., be solved by an appropriate numerical method (e.g., an iterative method), for example the fourth-order Runge-Kutta method.

In another embodiment, a (general) excitation signal, u(t), is used to drive the model. Initial guesses are made for the unknown parameters, R _e ,L _e ,Bl, K,M, and R _m . The system is solved, and the predicted current is compared to the measured current. To predict the driver’s linear parameters, a minimization problem is solved, with cost function where g = < R _e , L _e , Bl, K, M, R _m > is the vector of unknown parameters. The final solution provides the predicted velocity, V _p (ω). The normal velocity may, e.g., then be given by V ≈ V _p , wherein I _s is the measured current, 1(g) is the simulated current. The linear parameters are predicted by minimizing the difference between the measured and simulated current. The linear parameters do not modify the audio input channel. In other embodiments, other cost functions are employed

To estimate the sound pressure at the driver, the wave equation is solved to find the free- field transfer function (TF) from the center of the driver to measurement position m ₃ (see Fig. 4). Using this transfer function, the sound pressure at the source can be predicted from the measured sound pressure.

Different concepts are available for the acoustic modelling or simulation to generate a model, e.g., of the loudspeaker and the transfer function.

For example, the loudspeaker could be modeled in the free-field, with the assumption that all surfaces of the loudspeaker enclosure are acoustically hard. (More detailed models including boundary conditions of the room, and precise modelling of the loudspeakers surface and material properties would be possible).

Also specific situations that may be found in practical scenarios (e.g. positioning of the loudspeaker on a table, on or in a shelf, close to one, two, three boundary surfaces (e.g. close to wall, in a corner,...) may, e.g., be simulated and chosen on the actual situation in the listening environment (either automatic detection / selection, or by user). Also, in some of the embodiments, a simulation of the whole room, e.g. based on additional input data, is employed. (As an example, Fig. 13 depicts a loudspeaker on a surface/table).

A unit sound pressure may, e.g., be imposed at the driver, for a range of relevant input frequencies. The solution at position m ₃ is recovered. From this solution, a complex transfer function may, e.g., be computed as follows wherein P _src is the sound pressure imposed at the driver, and P _rec is the sound pressure received at position m ₃ . The required sound pressure is then given by P ≈ P _m3 /H.

In some of the embodiments, the above-described concepts are not limited to a usage of a single microphone. Instead, microphone arrays with a variable number of microphones in different arrangements (e.g. linear array, circular array, positioned at different surfaces of the loudspeakers enclosure) may, e.g., be used; see, for example, the embodiments illustrated by Fig. 12, Fig. 14, Fig. 15.

According to some of the embodiments, multiple recordings from the different microphones may, e.g., be employed. The one that gives the best recording in the present situation may, e.g., be selected. An average of all recorded signals to arrive at an overall better estimate compared to using only a single recording may, e.g., be calculated.

In some embodiments, the microphone may, for example, be an external microphone (e.g. also one of a mobile phone). For example, the exact model and position during measurement may, e.g., be known and may, e.g., be included in the simulation.

By driving the individual transducers (diaphragms) of a multi-driver-loudspeaker individually with a test signal, more information may, e.g., be gathered about the room (e.g. varying modal behavior).

A parameter model (e.g., a lumped parameter model) may, e.g., be used, and the system may, e.g., be continuously monitored. It may, e.g., be checked, if something in the setup or system behavior changes over time. E.g. a change in the position or environment could be detected.

According to another embodiment, the estimated velocity information (for example, the driver velocity) may, for example, be estimated once, e.g. during the design stage of the system.

For example, according to another embodiment, the estimated velocity information may, e.g., be stored within the apparatus 100.

Such an embodiment, may, for example, be based on the assumption that the magnitude profile of the estimated driver velocity (e.g., the frequency dependent magnitude of the velocity) does not change significantly between rooms, or in different positions within a room,

In an embodiment, the estimation during the design stage may, e.g. be performed by estimating in a laboratory environment the magnitude of the velocity in the complete/relevant (audio) frequency range for the specific loudspeaker or driver in response to e.g. an applied unit voltage or e.g. a known voltage.

The estimated velocity magnitude profile may then, e.g., be stored in a look-up table. Thus, in an embodiment, the estimated velocity information may, e.g., be stored in a look- up table which is stored within the apparatus 100. The estimation unit 110 may, e.g., be configured to derive the estimated velocity information from the look-up table.

In a linear audio system, a change in the driving voltage level (e.g., the audio input signal level) will result in a linearly proportional change in the driver velocity magnitude.

According to an embodiment, the estimation unit 110 may, e.g., be configured to derive the estimated velocity information from the look-up table using the driving voltage level as an input to the look-up table.

Thus, according to an embodiment, during runtime, the magnitude of the driver velocity could be determined from the driving voltage (and potentially a conversion factor) and the values stored in said look-up table, while the phase of the velocity could be estimated from the estimated pressure information, using the continuity of momentum.

In an embodiment, a kind of ‘health check’ of the system / drivers may, e.g., be performed. In some embodiments, it may, e.g., be monitored how the driver parameters change with time.

An apparatus comprising an estimation unit 110 is provided.

The estimation unit 110 is configured to estimate a first radiation resistance of each driver of one or more drivers of each loudspeaker of one or more loudspeakers as a first estimated radiation resistance before a first point in time; or is configured to estimate a first radiation impedance of each driver of the one or more drivers of each loudspeaker of the one or more loudspeakers as a first estimated radiation impedance before the first point in time, wherein said first estimated radiation impedance of said driver comprises estimated information on the first radiation resistance of said driver.

To estimate the first estimated radiation resistance or the first estimated radiation impedance of each driver of the one or more drivers of each loudspeaker of the one or more loudspeakers, the estimation unit 110 is configured to estimate the first estimated radiation resistance or the first estimated radiation impedance depending on first estimated sound pressure information indicating an estimation of sound pressure at said driver of said loudspeaker before the first point in time, and depending on first estimated velocity information indicating an estimation of a first driver velocity of said driver of said loudspeaker before the first point in time.

Furthermore, the estimation unit 110 is configured to estimate a second radiation resistance of each driver of the one or more drivers of each loudspeaker of the one or more loudspeakers as a second estimated radiation resistance after a second point in time; or is configured to estimate a second radiation impedance of each driver of the one or more drivers of each loudspeaker of the one or more loudspeakers as a second estimated radiation impedance after the second point in time, wherein said second estimated radiation impedance of said driver comprises estimated information on the second radiation resistance of said driver.

To estimate the second estimated radiation resistance or the second estimated radiation impedance of each driver of the one or more drivers of each loudspeaker of the one or more loudspeakers, the estimation unit 110 is configured to estimate the second estimated radiation resistance or the second estimated radiation impedance depending on second estimated sound pressure information indicating an estimation of sound pressure at said driver of said loudspeaker after the second point in time, and depending on second estimated velocity information indicating an estimation of a second driver velocity of said driver of said loudspeaker after the second point in time.

Furthermore, the estimation unit 110 is configured to determine and to output whether the apparatus is in a first state or whether the apparatus is in a second state depending on a radiation resistance difference indicating a difference between the second estimated radiation resistance and the first estimated radiation resistance, or depending on a radiation impedance difference indicating a difference between the second estimated radiation impedance and the first estimated radiation impedance.

The second state indicates that the apparatus is malfunctioning or that the apparatus has been relocated. The first state indicates that the apparatus is functioning and that the apparatus has not been relocated.

In an embodiment, the estimation unit 110 may, e.g., be configured to estimate the first estimated sound pressure information depending on captured first sound pressure information recorded by one or more microphones before the first point in time, and the estimation unit 110 may, e.g., be configured to estimate the second estimated sound pressure information depending on captured second sound pressure information recorded by one or more microphones after the second point in time. And/or the estimation unit 110 may, e.g., be configured to estimate the first estimated velocity information depending on a first current through a loudspeaker driver coil of said driver of said loudspeaker before the first point in time, and the estimation unit 110 may, e.g., be configured to estimate the second estimated velocity information depending on a second current through the loudspeaker driver coil of said driver of said loudspeaker after the second point in time.

In an embodiment, the estimation unit 110 may, e.g., be configured to determine the radiation resistance difference by determining a difference value indicating a difference between the second estimated radiation resistance and the first estimated radiation resistance; or is configured to determine the radiation impedance difference by determining a difference value indicating a difference between the second estimated radiation impedance and the first estimated radiation impedance. The estimation unit 110 may, e.g., be configured to determine that the apparatus is in the second state, if the difference value is greater than a threshold value. Moreover, the estimation unit 110 may, e.g., be configured to determine that the apparatus is in the first state, if the difference value is smaller than or equal to the threshold value.

In an embodiment, additionally other means, for example, from one or more gyroscopes, or other information that has been gathered from the pressure measurement, may, e.g., also be used as an indication that the device has been moved.

In some of the embodiments, a global EQ estimate from two different (or more) (spatially separated) loudspeakers may, e.g., be employed to get a better estimate of global EQ / of the room behavior.

In a particular embodiment, information gained from multiple loudspeakers may, e.g., be used to conduct modal equalization. Based on the actual position(s) of multiple loudspeakers and the estimated modal behavior, it may, for example, be checked, if an improvement in the reproduction in the modal frequency range can be achieved, and/or if one or more loudspeakers may, e.g., be used to compensate for modal effects of the other loudspeaker/room combinations.

In some of the embodiments, simulations that are used to estimate the sound pressure at the diaphragm may, for example, also include simulations of the surroundings to get better estimates. Those surroundings may, e.g., later be set by the user. Or, those surroundings may, e.g., be detected automatically. E.g. if the loudspeaker is positioned on a flat solid surface (e.g. a table), it will behave differently than in a bookshelf.

Fig. 10 illustrates a side view of an alternative loudspeaker layout with drivers/transducers at four sides according to an embodiment.

Fig. 11 illustrates a top view of an alternative loudspeaker layout with drivers/transducers at four sides according to an embodiment.

Fig. 12 illustrates an alternative loudspeaker layout being a soundbar-type with multiple microphones according to an embodiment.

Fig. 13 illustrates an example of a loudspeaker positioned on a surface (e.g. table) according to an embodiment.

Fig. 14 illustrates a top view of a loudspeaker showing potential positions for single or multiple microphones according to an embodiment.

Fig. 15 illustrates a side view of a loudspeaker showing potential positions for single or multiple microphones according to an embodiment.

Fig. 16 illustrates another side view of a loudspeaker showing potential positions for single or multiple microphones according to another embodiment.

In some embodiments it might be useful to place additional structures on the actual loudspeaker enclosure, as, e.g., means to diffuse the sound of some loudspeakers, e.g., loudspeakers firing upwards, by means of diffusors, spreaders, conic structures, diffusing cones, waveguides, etc., or other shapes to spread the sound in specific directions, e.g. horizontally, or in specific directions.

In such cases, the microphones can beneficially be placed on top of such structures, as exemplified in Fig. 16.

In the following, further embodiments are provided. In some of the embodiments, the performance of a loudspeaker in a room is controlled. The needed control parameters are (instead of being directly measured) estimated based on measurements of easily obtainable parameters. Those measured parameters are input parameters for at least one model that approximates the needed control parameters.

According to an embodiment, one of the models is an acoustic model, for example, an acoustic model to approximate the sound pressure at the diaphragm.

In an embodiment, one of the models is a simple plane wave approximation.

According to an embodiment, one of the models is a (detailed) wave based method, for example, a Finite Element Simulation. In an embodiment, a modelling of one or more properties of the specific loudspeaker may, e.g., be employed.

In an embodiment, the model to predict the sound pressure is a (simple) spherical wave approximation. For example, if the distance between a measurement point in front of a woofer, and the actual measurement point remote, for example, within a limited range of e.g. a few 10s of centimeters from the woofer is known, then the sound pressure at the woofer, e.g., in the low frequency region, can be computed/approximated from the remote measurement. The approximation that can be computed assumes sound to propagate as a spherical wave, and just takes into account the distance of the measurement point from the woofer. This approximation can be termed “spherical wave approximation”.

According to an embodiment, one of the models may, e.g., be an electro-mechanical model, for example, to approximate the velocity based on a current measurement.

In an embodiment, one of the easily obtainable parameters is a sound pressure measurement, which, e.g., does not have to be captured close to the diaphragm. For example, one or more microphones that conduct the sound pressure measurement can be (one or more) built in microphone(s) of a smart speaker, or, for example, a playback system that already features microphones for interaction, for example, with a voice-assistant.

According to an embodiment, each driver/transducer of a loudspeaker which comprises multiple drivers/transducers may, e.g., be used individually to select the best suited driver in the given situation, or, may, e.g., be used to calculate an average of all used drivers to enhance the result. In an embodiment, a specific test signal may, e.g., be used for calibrating the system. In another embodiment, instead, the played program material (e.g. music) may, e.g., be used for calibrating the system.

According to an embodiment, instead of a specific test signal, a special voice assistant phrase may, e.g., be used as test signal.

In an embodiment, the calibration may, e.g., be conducted at a specific instant in time (that, for example, may, e.g., be triggered by a user, e.g. after moving the loudspeaker).

According to another embodiment, instead of doing the calibration at a specific instant in time, the system may, e.g., conduct continuous adaption to the environment.

In an embodiment, the system may, e.g., only conduct a new calibration, if a change in the environment / setup position has been recognized.

Although some aspects have been described in the context of an apparatus, it is clear that these aspects also represent a description of the corresponding method, where a block or device corresponds to a method step or a feature of a method step. Analogously, aspects described in the context of a method step also represent a description of a corresponding block or item or feature of a corresponding apparatus. Some or all of the method steps may be executed by (or using) a hardware apparatus, like for example, a microprocessor, a programmable computer or an electronic circuit. In some embodiments, one or more of the most important method steps may be executed by such an apparatus.

Depending on certain implementation requirements, embodiments of the invention can be implemented in hardware or in software or at least partially in hardware or at least partially in software. The implementation can be performed using a digital storage medium, for example a floppy disk, a DVD, a Blu-Ray, a CD, a ROM, a PROM, an EPROM, an EEPROM or a FLASH memory, having electronically readable control signals stored thereon, which cooperate (or are capable of cooperating) with a programmable computer system such that the respective method is performed. Therefore, the digital storage medium may be computer readable. Some embodiments according to the invention comprise a data carrier having electronically readable control signals, which are capable of cooperating with a programmable computer system, such that one of the methods described herein is performed.

Generally, embodiments of the present invention can be implemented as a computer program product with a program code, the program code being operative for performing one of the methods when the computer program product runs on a computer. The program code may for example be stored on a machine readable carrier.

Other embodiments comprise the computer program for performing one of the methods described herein, stored on a machine readable carrier.

In other words, an embodiment of the inventive method is, therefore, a computer program having a program code for performing one of the methods described herein, when the computer program runs on a computer.

A further embodiment of the inventive methods is, therefore, a data carrier (or a digital storage medium, or a computer-readable medium) comprising, recorded thereon, the computer program for performing one of the methods described herein. The data carrier, the digital storage medium or the recorded medium are typically tangible and/or non- transitory.

A further embodiment of the inventive method is, therefore, a data stream or a sequence of signals representing the computer program for performing one of the methods described herein. The data stream or the sequence of signals may for example be configured to be transferred via a data communication connection, for example via the Internet.

A further embodiment comprises a processing means, for example a computer, or a programmable logic device, configured to or adapted to perform one of the methods described herein.

A further embodiment comprises a computer having installed thereon the computer program for performing one of the methods described herein.

A further embodiment according to the invention comprises an apparatus or a system configured to transfer (for example, electronically or optically) a computer program for performing one of the methods described herein to a receiver. The receiver may, for example, be a computer, a mobile device, a memory device or the like. The apparatus or system may, for example, comprise a file server for transferring the computer program to the receiver.

In some embodiments, a programmable logic device (for example a field programmable gate array) may be used to perform some or all of the functionalities of the methods described herein. In some embodiments, a field programmable gate array may cooperate with a microprocessor in order to perform one of the methods described herein. Generally, the methods are preferably performed by any hardware apparatus.

The apparatus described herein may be implemented using a hardware apparatus, or using a computer, or using a combination of a hardware apparatus and a computer. The methods described herein may be performed using a hardware apparatus, or using a computer, or using a combination of a hardware apparatus and a computer.

The above described embodiments are merely illustrative for the principles of the present invention. It is understood that modifications and variations of the arrangements and the details described herein will be apparent to others skilled in the art. It is the intent, therefore, to be limited only by the scope of the impending patent claims and not by the specific details presented by way of description and explanation of the embodiments herein.