Method for operating a loudspeaker

Abstract

The invention relates to a method for operating a loudspeaker, in particular a moving-coil loudspeaker, in which a signal used to generate sound is linearized by the loudspeaker. According to the invention, the signal is filtered in a frequency-dependent manner, and only a filtered portion of the signal is linearized. Advantageously, the signal is filtered by means of a high-pass and / or a low-pass filter. In one embodiment of the invention, only the portion of the signal output by the low-pass filter is linearized.

Description

[0001] The invention relates to a method for operating a loudspeaker, in particular a moving-coil loudspeaker, in which a useful signal is linearized to generate sound by means of the loudspeaker.

[0002] It is known from experience to linearize the drive parameters of moving-coil loudspeakers using digital signal processing in order to improve the sound. Linearization is complex and computationally intensive, as various loudspeaker parameters are used to perform the necessary calculations.

[0003] The invention is based on the objective of reducing the computational effort required for digital signal processing.

[0004] According to the invention, this problem is solved by filtering the useful signal in a frequency-dependent manner and linearizing only a filtered portion of the useful signal.

[0005] To operate the aforementioned mathematical model and perform the linearization, various parameters of the loudspeaker must be considered, e.g., an electrical voice coil resistance RE, oscillating mass m, mechanical friction, suspension stiffness KME, voice coil inductance Le, and electromagnetic force factor Bl.

[0006] Nonlinearities of each parameter are considered in higher-order polynomials, e.g., according to the following formula using a 4th-order polynomial: BL(x)=Bl0+Bl1⋅x+Bl2⋅x2+Bl3⋅x3+Bl4⋅x4

[0007] If each of the aforementioned parameters is considered to be of the 4th order, the present example results in a 24th-order computational problem. The computational effort depends on the number of parameters that need to be tracked and the required sampling rate for determining the parameters. Thanks to the invention, the computational effort can be significantly reduced, since linearization only needs to be performed on a portion of the useful signal.

[0008] In one embodiment of the invention, the useful signal is filtered by means of a high-pass and / or a low-pass filter, preferably a Linwitz-Riley filter in each case.

[0009] In a particularly preferred embodiment of the invention, only a portion of the useful signal output by the low-pass filter is linearized. In particular, a portion of the useful signal output by the high-pass filter is not linearized.

[0010] It has surprisingly been shown that a high-frequency component of the signal does not necessarily need to be linearized. This is because its reproduction does not require large diaphragm excursions, and therefore linearization is not essential. Since only low sampling rates are required for the low-frequency component, the computational effort is further reduced.

[0011] Advantageously, the low-pass filter has a cutoff frequency in a range of 100 - 1200 Hz, preferably 250 - 1100 Hz, particularly preferably 500 - 1000 Hz.

[0012] In a further embodiment of the invention, the portion of the useful signal output by the high-pass filter is delayed for superposition with the portion of the useful signal output by the low-pass filter. Advantageously, the portion of the useful signal output by the high-pass filter is delayed in such a way as to compensate for any delay resulting from processing of the portion of the useful signal output by the low-pass filter. Preferably, the portion of the useful signal output by the high-pass filter is delayed by means of a delay element, preferably a digital one.

[0013] In one embodiment of the invention, the sampling rate F1 of the portion of the useful signal output by the low-pass filter is reduced by means of a data rate converter, preferably to a sampling rate F2.

[0014] Advantageously, the sampling rate of the portion of the useful signal output by the low-pass filter is reduced by means of the data rate converter before the portion of the useful signal is linearized. Preferably, the sampling rate of the linearized portion of the useful signal is increased to the sampling rate F1 by means of a data rate converter.

[0015] In a preferred embodiment of the invention, the portion of the useful signal that is output by the high-pass filter and optionally further processed by means of the delay, and the portion of the useful signal that is output by the low-pass filter and optionally further processed, are superimposed to form a digital output signal.

[0016] Advantageously, the digital output signal is processed by means of an amplifier that includes a digital-to-analog converter.

[0017] In one embodiment of the invention, a measured quantity that can be used for linearization, in particular current and / or voltage, with which the loudspeaker is operated and / or which is generated by an analog signal output by the amplifier, is measured by means of a sensor. Alternatively, it would be conceivable to determine the displacement of a diaphragm of the loudspeaker or the sound pressure generated by the loudspeaker.

[0018] It is advantageous to delay a measurement signal from the sensor for use in linearization and / or to reduce the sampling rate of the analog signal to the sampling rate F2 using a data rate converter.

[0019] The invention further relates to a computer program comprising instructions which, when the program is executed by a computer, cause it to perform the method described above.

[0020] The computer program is expediently stored on a digital signal processor, a microcontroller, a data carrier, preferably RAM, ROM, CD or the like, or a device, in particular a personal computer, a device with an embedded processor, preferably a digital signal processor or a microcontroller, a computer embedded in a device, a smartphone, a computer program or a sequence of signals suitable for transmission via a computer network, in particular the Internet, representing data.

[0021] The device in question may be an amplifier or a loudspeaker, or the device may include an amplifier or a loudspeaker.

[0022] The invention further relates to a data carrier signal which transmits the aforementioned computer program.

[0023] Furthermore, the invention relates to a processor, in particular a digital signal processor or microcontroller, on which the computer program is stored, an audio amplifier comprising a digital signal processor or a microcontroller on which the computer program is stored, and / or a sound reinforcement unit comprising an audio amplifier comprising a digital signal processor or a microcontroller on which the computer program is stored, and a loudspeaker connected to the audio amplifier.

[0024] In a further development of the invention, the invention relates to a motor vehicle, in particular an automobile, which is equipped with a sound system.

[0025] The invention is explained in more detail below with reference to an exemplary embodiment and the accompanying drawings relating to that embodiment. Fig.Figure 1 schematically shows steps of the method according to the invention.

[0026] A useful signal Audio_in x i (n) is filtered by means of a high-pass filter 1 and a low-pass filter 2, both of which can be formed by Linwitz-Riley filters. The cutoff frequencies of the two filters 1, 2 are, for example, 800 Hz.

[0027] The portion of the useful signal output by the high-pass filter 1 is delayed by means of a delay element 3 for superposition with the portion of the useful signal output by the low-pass filter 2, such that a delay resulting from processing of the portion of the useful signal output by the low-pass filter 2 is compensated.

[0028] A sampling rate F1 of the portion of the useful signal output by the low-pass filter 2 is reduced to a sampling rate F2 by means of a data rate converter 4 before the portion of the useful signal is linearized by means of a linearization unit 5. For example, F1 can be 48 kHz and F2 can be 6 kHz. The sampling rate of the linearized portion of the useful signal is then increased to the original sampling rate F1 by means of a data rate converter 6.

[0029] The portion of the useful signal output by the high-pass filter 1 after delay by the delay element 3, and the portion of the linearized and delayed useful signal output from the low-pass filter 2, are superimposed in a mixer 9 to form a digital output signal. The digital output signal is processed by an amplifier 10, which includes a digital-to-analog converter.

[0030] For linearization, the current with which a loudspeaker 12 is operated is measured using a current sensor 11. A measurement signal from the current sensor 11 is delayed by the linearization unit 5 by means of a further delay element 8 and reduced to the sampling rate F2 by means of a data rate converter 7, so that the information obtained from the current sensor 11 can be used for linearization.

Claims

[1] Method for operating a loudspeaker, in particular a moving-coil loudspeaker, in which a useful signal is linearized to produce sound by means of the loudspeaker, characterized by , that the useful signal is filtered depending on the frequency and only a filtered portion of the useful signal is linearized. [2] Method according to claim 1, characterized by that the useful signal is filtered using a high-pass and / or a low-pass filter, preferably a Linwitz-Riley filter. [3] Method according to claim 1 or 2, characterized by , that only a portion of the useful signal is linearized, which is output by the low-pass filter. [4] Method according to claim 2 or 3, characterized by that the low-pass filter has a cutoff frequency in a range of 100 - 1200 Hz, preferably 250 - 1100 Hz, particularly preferably 500 - 1000 Hz. [5] Method according to any one of claims 2 to 4, characterized by, that a portion of the useful signal output by the high-pass filter is not linearized. [6] Method according to any one of claims 2 to 4, characterized by , that the portion of the useful signal output by the high-pass filter is delayed for superposition with the portion of the useful signal output by the low-pass filter. [7] Method according to any one of claims 2 to 6, characterized by , that the portion of the useful signal output by the high-pass filter is delayed in such a way as to compensate for any delay caused by processing of the portion of the useful signal output by the low-pass filter. [8] Method according to any one of claims 2 to 7, characterized by , that the portion of the useful signal output by the high-pass filter is delayed by means of a delay element, preferably digital. [9] Method according to any one of claims 2 to 8, characterized by , that a sampling rate F1 of the portion of the useful signal output by the low-pass filter is reduced by means of a data rate converter, preferably to a sampling rate F2. [10] Method according to any one of claims 2 to 9, characterized by , that the sampling rate of the portion of the useful signal output by the low-pass filter is reduced by means of the data rate converter before the portion of the useful signal is linearized. [11] Method according to any one of claims 1 to 10, characterized by , that the sampling rate of the linearized portion of the useful signal is increased to the sampling rate F1 by means of a data rate converter. [12] Method according to any one of claims 1 to 11, characterized by, that the portion of the useful signal that is output by the high-pass filter and possibly further processed by means of the delay, and the portion of the useful signal that is output by the low-pass filter and possibly further processed, are superimposed to form a digital output signal. [13] Method according to claim 12, characterized by , that the digital output signal is processed by means of an amplifier which includes a digital-to-analog converter. [14] Method according to claim 13, characterized by , that a current and / or voltage, with which the loudspeaker is operated and / or which is generated by an analog signal output by the amplifier, is measured by means of a sensor. [15] Method according to claim 14, characterized by, that a measurement signal from the sensor is delayed for use in linearization by means of a delay and / or the sampling rate of the analog signal is reduced to the sampling rate F2 by means of a data rate converter. [16] Computer program comprising instructions which, when the program is executed by a computer, cause it to execute the method according to any one of claims 1 to 15. [17] Computer program according to claim 16, characterized by that the computer program is stored on a digital signal processor, a data carrier, preferably RAM, ROM, CD or the like, or a device, in particular a personal computer, a device with an embedded processor, preferably a digital signal processor, a computer embedded in a device, a smartphone, or is a sequence of signals suitable for transmission over a computer network, in particular the Internet, representing data. [18] Computer program according to claim 16, characterized by that the device is an amplifier and / or a loudspeaker, or that the device includes an amplifier and / or a loudspeaker. [19] Data carrier signal which transmits the computer program according to claim 16. [20] Processor, in particular a digital signal processor or a microcontroller on which the computer program according to claim 16 is stored. [21] Audio amplifier comprising a processor, in particular a digital signal processor or a microcontroller, on which the computer program according to claim 16 is stored. [22] Sound reinforcement unit comprising an audio amplifier comprising a digital signal processor on which the computer program according to claim 16 is stored and a loudspeaker connected to the audio amplifier. [23] Motor vehicle, in particular automobile, which is equipped with a sound system according to claim 22.

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