阅读说明：本技术 用于处理音频信号的设备 (Apparatus for processing audio signals ) 是由 B·塞拉克 于 2019-10-21 设计创作，主要内容包括：一种用于处理包括多个样本的音频信号、特别是以便在该音频信号中生成缺失的低频分量谐波的设备,包括至少一个音频处理装置,被配置为：在音频信号的时间相关表示中、特别是在音频信号的半波表示中处理音频信号；确定在音频信号的第一过零点和另一个过零点之间的间隔；确定在所述间隔中的第一组采样点,第一组采样点包括在所述间隔中的第一位置处的多个采样点；确定在所述间隔中的第二组采样点,第二组采样点包括在所述间隔中的第二位置处的多个采样点；基于音频信号修改规则,通过以下方式在所述间隔中修改所述音频信号,即,改变在所述间隔中的第一组采样点中的采样点的位置,使得第一组采样点中的每个采样点从其在第一组采样点中的相应第一位置改变为其在第二组采样点中的相应第二位置；将经修改的音频信号间隔应用到原始音频信号的相应间隔,以便生成经修改的音频信号。(An apparatus for processing an audio signal comprising a plurality of samples, in particular in order to generate missing low frequency component harmonics in the audio signal, comprising at least one audio processing device configured to: processing the audio signal in a time-dependent representation of the audio signal, in particular in a half-wave representation of the audio signal; determining an interval between a first zero-crossing and a further zero-crossing of the audio signal; determining a first set of sample points in the interval, the first set of sample points comprising a plurality of sample points at a first position in the interval; determining a second set of sampling points in the interval, the second set of sampling points comprising a plurality of sampling points at a second location in the interval; modifying the audio signal in the interval based on an audio signal modification rule by changing the positions of the sample points in a first set of sample points in the interval such that each sample point in the first set of sample points changes from its respective first position in the first set of sample points to its respective second position in the second set of sample points; the modified audio signal intervals are applied to corresponding intervals of the original audio signal to generate a modified audio signal.)

1. An apparatus for processing an audio signal comprising a plurality of samples, in particular in order to generate missing low frequency component harmonics in the audio signal, the apparatus comprising at least one audio processing device configured to:

-processing an audio signal in a time-dependent representation of the audio signal, in particular in a half-wave representation of the audio signal;

-determining an interval between a first zero-crossing and a further zero-crossing of the audio signal;

-determining a first set of sample points in the interval, the first set of sample points comprising a plurality of sample points at a first position in the interval;

-determining a second set of sampling points in the interval, the second set of sampling points comprising a plurality of sampling points at a second position in the interval;

-modifying the audio signal in the interval based on an audio signal modification rule by: changing the position of the sampling points in the first set of sampling points in the interval such that each sampling point in the first set of sampling points changes from its respective first position in the first set of sampling points to its respective second position in the second set of sampling points;

-applying the modified audio signal interval to a corresponding interval of the original audio signal in order to generate a modified audio signal.

2. The apparatus of claim 1, wherein the audio processing device is configured to: modifying the audio signal based on an audio signal modification rule that specifies a defined variation of positions of sample points in the first set of sample points in the interval such that each sample point in the first set of sample points changes from its respective first position in the first set of sample points to its respective second position in the second set of sample points.

3. The apparatus of claim 1 or 2, wherein the audio signal modification rule specifies a defined change in the positions of the sampling points in the first set of sampling points in the interval such that each sampling point in the first set of sampling points changes from its respective first position in the first set of sampling points to its respective second position in the second set of sampling points such that the sampling points in the second set of sampling points are equally spaced.

4. The device according to claim 2 or 3, wherein the audio signal modification rule is or comprises a mapping function, in particular a monotonic mapping function, configured to map input sample points of the first set of samples having respective first positions to output sample points of the second set of sample points having respective second positions.

5. The apparatus according to any of claims 2 to 4, wherein the audio signal modification rule is or comprises a tilt function configured to tilt a zero-crossing tangent of the audio signal in a clockwise or counter-clockwise direction.

6. Apparatus according to any one of the preceding claims, wherein the audio signal processable by or by the audio processing means has a particular raw waveform, thereby

The audio processing device is configured to modify the particular original waveform of the audio signal into at least one target waveform of the audio signal.

7. The apparatus of claim 6, wherein the audio processing device is configured to: modifying the particular original waveform of the audio signal based on the audio signal modification rule or an audio signal modification rule specifying a defined change in waveform of the audio signal from its original waveform to at least one target waveform of the audio signal.

8. The apparatus according to claim 6 or 7, wherein the target waveform is a symmetric waveform, in particular a rectangular waveform, a triangular waveform or a needle-like waveform; or an asymmetrical waveform, in particular a sawtooth waveform, preferably a straight or curved falling or rising sawtooth waveform.

9. The apparatus according to any of the preceding claims, wherein the audio processing device is configured to apply a skip rule or a skip factor according to which at least one zero-crossing between a first zero-crossing and another zero-crossing is not considered for determining the interval between the first zero-crossing and the another zero-crossing of the audio signal.

10. The apparatus of any preceding claim, further comprising at least one filter means, in particular a low pass filter means, arranged to apply at least one filtering rule to the audio signal before the audio signal is processed by the audio processing means, and/or at least one filter means, in particular a low pass filter means, arranged to apply at least one filtering rule to the audio signal after the audio signal is processed by the audio processing means.

11. Apparatus according to any preceding claim, comprising a first audio processing means and at least one further audio processing means arranged in a parallel arrangement.

12. The apparatus of claim 11, wherein the first audio processing device is configured to modify the audio signal based on a first audio signal modification rule by: changing the positions of the sample points in the first set of sample points in the interval such that each sample point in the first set of sample points changes from its respective first position in the first set of sample points to its respective second position in the second set of sample points, and

the at least one further audio processing apparatus is configured to modify the audio signal based on at least one further audio signal modification rule by: changing the position of the sampling points in the first set of sampling points in the interval such that each sampling point in the first set of sampling points changes from its respective first position in the first set of sampling points to its respective second position in the second set of sampling points.

13. The apparatus of claim 12, wherein the first modification rule of the first audio processing device specifies a defined change in the waveform of the audio signal from its original waveform to at least one first target waveform of the audio signal, and

the at least one further modification rule of the at least one further audio processing device specifies a defined change of the waveform of the audio signal from its original waveform to at least one further target waveform of the audio signal.

14. The apparatus according to claim 13, wherein the first target waveform of the audio signal is opposite to the at least one further target waveform of the audio signal.

15. The apparatus according to any of the preceding claims, wherein the first audio processing device is configured to apply a first skip rule or a skip factor according to which at least one zero-crossing between a first zero-crossing and another zero-crossing is not taken into account for determining the interval between the first zero-crossing and the another zero-crossing of the audio signal, and

the at least one further audio processing device is configured to apply at least one further skip rule or skip factor according to which at least one zero crossing between a first zero crossing and a further zero crossing is not taken into account for determining the interval between the first zero crossing and the further zero crossing of the audio signal.

16. The apparatus according to any of the preceding claims, wherein the at least one audio processing device is configured to determine the number of sample points between the first zero-crossing and the further zero-crossing so as to be the same as the number of sample points in the respective interval in the original audio signal.

17. A device for outputting an audio signal, in particular in a vehicle cabin, comprising a plurality of samples, in particular in order to generate a missing low frequency component in the audio signal, the device comprising:

-at least one audio processing device configured to:

-processing an audio signal comprising a plurality of evenly spaced sampling points in a time-dependent representation of the audio signal, in particular in a half-wave representation of the audio signal;

-determining an interval between a first zero-crossing and a further zero-crossing of the audio signal;

-determining a first set of sample points in said interval, said first set of sample points comprising a plurality of sample points at a first position in said interval;

-determining a second set of sampling points in said interval, said second set of sampling points comprising a plurality of sampling points at second locations in said interval;

-modifying the audio signal in the interval based on an audio signal modification rule by: changing the position of the sampling points in the first set of sampling points in the interval such that each sampling point in the first set of sampling points changes from its respective first position in the first set of sampling points to its respective second position in the second set of sampling points;

-applying the modified audio signal interval to a corresponding interval of the original audio signal in order to generate a modified audio signal; and

-at least one audio output device configured to output the modified audio signal, in particular in a vehicle cabin.

18. A method for processing an audio signal comprising a plurality of samples, in particular in order to generate a missing low frequency component in the audio signal, the method comprising:

-processing an audio signal comprising a plurality of evenly spaced sampling points in a time-dependent representation of the audio signal, in particular in a half-wave representation of the audio signal;

-determining an interval between a first zero-crossing and a further zero-crossing of the audio signal;

-determining a first set of sample points in the interval, the first set of sample points comprising a plurality of sample points at a first position in the interval;

-determining a second set of sampling points in the interval, the second set of sampling points comprising a plurality of sampling points at a second position in the interval;

-modifying the audio signal in the interval based on an audio signal modification rule by: changing the position of the sampling points in the first set of sampling points in the interval such that each sampling point in the first set of sampling points changes from its respective first position in the first set of sampling points to its respective second position in the second set of sampling points;

-applying the modified audio signal interval to a corresponding interval of the original audio signal in order to generate a modified audio signal.

19. Method for outputting an audio signal, in particular in a vehicle cabin, comprising a plurality of samples, in particular in order to generate a missing low frequency component in the audio signal, the method comprising:

-processing an audio signal comprising a plurality of evenly spaced sampling points in a time-dependent representation of the audio signal, in particular in a half-wave representation of the audio signal;

-determining an interval between a first zero-crossing and a further zero-crossing of the audio signal;

-determining a first set of sample points in the interval, the first set of sample points comprising a plurality of sample points at a first position in the interval;

-determining a second set of sampling points in the interval, the second set of sampling points comprising a plurality of sampling points at a second position in the interval;

-modifying the audio signal in the interval based on an audio signal modification rule by: changing the position of the sampling points in the first set of sampling points in the interval such that each sampling point in the first set of sampling points changes from its respective first position in the first set of sampling points to its respective second position in the second set of sampling points;

-applying the modified audio signal interval to a corresponding interval of the original audio signal in order to generate a modified audio signal;

-outputting the modified audio signal, in particular in a vehicle cabin.

Technical Field

The invention relates to an apparatus for processing an audio signal comprising a plurality of samples, in particular in order to generate missing low frequency component harmonics in the audio signal.

Background

Processing of audio signals on audio output devices such as mobile electronic devices, mobile loudspeakers, etc., which have a poor low frequency response due to constructional and/or physical limitations, i.e. in particular by reproducing audio signals, is a known challenge in the field of audio signal processing.

In view of this challenge, known non-linear audio signal processing devices (e.g. known as "Maxxbass" or "Dirac bases") allow for (substantially) non-linear distortion based Bass enhancement. The respective audio signal processing means typically comprise a sample-wise weighting of an audio signal comprising a plurality of samples having a non-linear characteristic. The respective audio signal processing means typically implement a "horizontal distortion" of the audio signal by modifying the amplitude of the samples.

Thus, the level of the generated harmonics and thus the magnitude of the acoustically perceptible virtual bass boost is highly dependent on the audio signal level. In addition, the resulting harmonic instability needs to be mitigated by determining a loudness estimate and applying an automatic gain control stage (AGC stage), which often presents additional difficulties.

Therefore, there is a need for an improved method for processing an audio signal comprising a plurality of samples, in particular in order to generate missing low frequency component harmonics in the audio signal.

Disclosure of Invention

It is an object of the invention to provide an improved device for processing an audio signal comprising a plurality of samples, in particular in order to generate low frequency component harmonics, in particular missing harmonics, in the audio signal.

This object is achieved by a device for processing an audio signal comprising a plurality of samples, in particular in order to generate low frequency component harmonics in the audio signal, according to claim 1. The claims dependent on claim 1 relate to possible embodiments of the device according to claim 1.

A first aspect of the invention relates to an apparatus for processing an audio signal comprising a plurality of samples, in particular in order to generate low frequency component harmonics in the audio signal.

The device is generally applicable to a wide range of audio applications. The apparatus is generally applicable to any audio application in which a poor low frequency response is given, for example due to constructional and/or physical limitations of the audio output element (e.g. speaker). In other words, the apparatus may be generally applied to any audio application in which virtual bass enhancement is used to compensate for missing harmonics of low frequency components in an audio signal that may also be considered or represented as bass components due to structural and/or physical limitations of the audio output element (e.g., speaker).

An exemplary audio application of the apparatus is a mobile device application or a portable device application. Thus, the apparatus may be installed in a mobile device or a portable device, such as a mobile computer, a smart phone, a tablet computer, a mobile speaker, etc.

The preferred audio application of the device is a car audio application. Thus, the device may be installed in a vehicle or automobile, respectively. Thus, the device may be provided as, or may form part of, a vehicle audio system or a car audio system, respectively. In automotive applications, the apparatus may allow compensation for missing low frequency component harmonics in an audio signal due to structural and/or physical limitations of an audio output element (e.g., speaker) disposed in a vehicle or automobile, respectively.

The device, regardless of its application, may be embodied in hardware and/or software.

The apparatus comprises at least one hardware and/or software embodied audio processing device.

The at least one audio processing device is configured to: an input audio signal comprising a plurality of samples is processed in a time-dependent representation of the input audio signal, in particular in a half-wave representation of the input audio signal. Time-dependent representation of an input audio signal: typically or comprising a time-dependent representation of spaced sample points of the input audio signal, more particularly a time-dependent representation of non-uniformly spaced sample points of the input audio signal. The time-dependent representation of the input audio signal may comprise a representation of a graphical function (curve) interconnecting the sample points of the input audio signal along a time axis, i.e. typically the x-axis representing the samples of the input audio signal, or a corresponding graphical function (curve) interconnecting the sample points of the input audio signal along a time axis, i.e. typically the x-axis representing the samples of the input audio signal. For example, the corresponding graphical function may be determined by interpolation of sample points of the input audio signal. Thus, the audio processing apparatus is configured to generate a time-dependent representation of the input audio signal, in particular a half-wave representation of the input audio signal, from the input audio signal comprising a plurality of samples. Thus, during operation of the device, the audio processing means process the respective input audio signal in a time-dependent representation of the input audio signal, in particular in a half-wave representation of the input audio signal. Thus, during operation of the device, the audio processing means generate a time-dependent representation of the input audio signal, in particular a half-wave representation of the input audio signal, from the respective input audio signal.

The audio processing apparatus is further configured to: an interval between a first zero-crossing and another zero-crossing of the input audio signal in a time-dependent representation of the input audio signal is determined. Thus, the audio processing apparatus is configured to: the zero-crossing points of a time-dependent representation of the input audio signal, i.e. the positions at which the respective graphical function interconnecting the sampling points of the input audio signal crosses the time axis in the time-dependent representation, are analyzed and, based on the determination of the respective zero-crossing points, the interval between a first zero-crossing point, i.e. a first position at which the respective graphical function interconnecting the sampling points of the input audio signal in the time-dependent representation crosses the time axis for the first time, and another zero-crossing point (or a second zero-crossing point), i.e. another position at which the respective graphical function interconnecting the sampling points of the input audio signal in the time-dependent representation crosses the time axis for another (or a second) time, is determined. Thus, during operation of the device, the audio processing means analyze respective zero-crossings of the time-dependent representation of the input audio signal, i.e. positions in the time-dependent representation of the input audio signal at which the respective graph function interconnecting the sampling points of the input audio signal crosses the time axis, and determine an interval between the respective first zero-crossing and the respective further zero-crossing (or second zero-crossing) based on the determination of the respective zero-crossing.

The respective first zero-crossing and the further zero-crossing may be directly consecutive zero-crossings. However, it is also possible that the respective first zero-crossing and the further zero-crossing are not directly consecutive zero-crossings, but are indirectly consecutive zero-crossings, such that at least one zero-crossing is located between the respective first zero-crossing and the respective further zero-crossing. Thus, the respective interval may extend between two directly consecutive zero-crossings of the time-dependent representation of the input audio signal, or the respective interval may extend between two indirectly consecutive zero-crossings of the time-dependent representation of the input audio signal.

The at least one audio processing device is further configured to: a first set of sample points in the determined interval is determined, the first set of sample points including a plurality of sample points at a first position in the interval. Thus, during operation of the device, the audio processing means determines a first set of sampling points in the interval, the first set of sampling points comprising a plurality of sampling points at a first position in the interval. The positions of the sample points in the first set of sample points in the interval typically represent the original positions of the sample points of the input audio signal in the interval as given in the time-dependent representation of the input audio signal. In other words, the positions of the sampling points in the first set of sampling points typically correspond to the original positions of the sampling points of the input audio signal in the interval as given in a time-dependent representation of the input audio signal obtained by processing the input audio signal.

The at least one audio processing device is further configured to: a second set of sample points in the determined interval is determined, the second set of sample points including a plurality of sample points at a second location in the interval. Thus, during operation of the apparatus, the at least one audio processing device determines a second set of sampling points in the interval, the second set of sampling points comprising a plurality of sampling points at a second position in the interval. The positions of the sampling points in the second set of sampling points typically represent the target positions of the sampling points of the input audio signal in the interval and are therefore offset from the original positions of the sampling points of the input audio signal in the interval as given in the time-dependent representation of the input audio signal. In other words, the positions of the sampling points P in the second set of sampling points in the interval typically correspond to positions of the position offsets of the sampling points in the first set of sampling points in the interval as given in the time-dependent representation of the input audio signal.

The number of sample points in the first set of sample points is typically equal to the number of sample points in the second set of sample points.

The at least one audio processing device is further configured to: modifying the input audio signal in the interval based on the audio signal modification rule by: the positions of the sampling points in the first set of sampling points in the interval are changed such that each sampling point in the first set of sampling points changes from its respective first position in the first set of sampling points to its respective second position in the second set of sampling points. Thus, during operation of the device, the at least one audio processing means changes the positions of the sampling points in the first set of sampling points in the interval based on, i.e. using, the audio signal modification rule, such that each sampling point in the first set of sampling points changes from its respective first position in the first set of sampling points to its respective second position in the second set of sampling points. Thus, the audio signal modification rule may specify a change in the position of the sampling points in the interval such that the position of each sampling point changes from its initial position in the first set of sampling points to its target position in the second set of sampling points. Thus, the modification rule may also specify an offset between the position of the respective sample point in the first set of sample points (i.e., before the position of the respective sample point has changed) and the changed position of the respective sample point in the second set of sample points (i.e., after the position of the respective sample point has changed).

The at least one audio processing device is further configured to: the modified audio signal intervals are applied to corresponding intervals of the original input audio signal to generate a modified audio signal. Thus, during operation of the apparatus, the at least one audio processing device applies the modified audio signal interval to a corresponding interval of the original input audio signal in order to generate a modified audio signal. The modified audio signal is acoustically perceptible or acoustically perceptible as if the original input audio signal would include generated harmonics of the low frequency components. The modified audio signal is typically the input audio signal with a level that is constant such that no automatic gain control stage needs to be applied.

The modified audio signal may be output in an acoustic environment (e.g., a vehicle cabin) via an audio output device that includes one or more audio output elements, such as speakers.

The audio processing device may be provided with computer readable instructions which, when executed by the processing unit of the audio processing device, enable the audio processing device to implement the above processing, determining, modifying and applying aspects specified above.

The apparatus allows for efficient principles of generating low frequency component harmonics of an input audio signal with relatively low complexity to thereby make it suitable for real-time applications.

Thus, as can be seen from the above description of the operation of the at least one audio processing device, the at least one audio processing device is configured to: an input audio signal having a plurality of samples is resampled, particularly on a non-uniformly spaced basis and particularly on a uniformly spaced basis, the samples being re-spread by changing the positions of the sampling points of the first set of sampling points such that each sampling point of the first set of sampling points changes from its respective first position in the first set of sampling points to its respective second position in the second set of sampling points.

By way of example only, an input audio signal representing a positive purely sinusoidal half-wave is resampled at a low sample point density at the beginning of the half-wave, while a higher and higher sample point density towards the end of the half-wave may result in a waveform of the audio signal that resembles a falling sawtooth waveform. If the next negative half-wave is resampled with an inverse sample point density, the resulting audio signal will have the same fundamental frequency as the original sinusoidal half-wave but a harmonic pattern similar to a sawtooth half-wave.

The at least one audio processing device may be configured to: the number of sampling points between the first zero-crossing and the at least one further zero-crossing is determined such that it is the same as the number of sampling points in the corresponding interval in the original input audio signal. The number of sample points between the first zero crossing and the at least one further zero crossing is determined such that the same number of sample points in the corresponding interval in the original input audio signal generally has a positive influence on the generation of the low frequency component harmonics.

The at least one audio processing device may be configured to: the audio signal is modified based on an audio signal modification rule that specifies a definable or defined change in the position of the sampling points in the first set of sampling points in the interval such that each sampling point in the first set of sampling points changes from its respective first position in the first set of sampling points to its respective second position in the second set of sampling points.

The audio signal modification rule may particularly specify a defined variation of the positions of the sampling points in the first set of sampling points in the interval such that each sampling point in the first set of sampling points changes from its respective first position in the first set of sampling points to its respective second position in the second set of sampling points such that the sampling points in the second set of sampling points are equally or uniformly spaced. Thus, the audio processing apparatus may be configured to: the samples are again equally or uniformly spread by changing the positions of the sampling points in the first set of sampling points, on the premise that the positions of the sampling points in the second set of sampling points are equally or uniformly spaced, so that each sampling point in the first set of sampling points changes from its respective first position in the first set of sampling points to its respective second position in the second set of sampling points.

The audio signal modification rule may be or may comprise a mapping function, in particular a monotonic mapping function, configured to map input sample points of the first set of sample points having respective first positions to output sample points of the second set of sample points having respective second positions. The mapping function may specifically map input sample points within a predefinable or predefined range (e.g. within a range of [0,1 ]) to output sample points within a predefinable or predefined range. Thus, the at least one audio processing device may be configured to map the position of each sample point of the first set of sample points to a defined position in the second set of sample points based on the respective mapping function. The mapping function may specifically allow for evenly spaced positions of the sampling points of the second set of sampling points. The mapping function allows to consistently affect the acoustically perceptible properties of the modified audio signal.

Additionally or alternatively, the audio signal modification rule may be or comprise a tilt function configured to tilt a zero-crossing tangent of the input audio signal in a clockwise or counter-clockwise direction. Thus, the at least one audio processing apparatus may be configured to tilt the zero-crossing tangents of the input audio signal, i.e. the tangents representing in the respective zero-crossings of the respective graphical functions (curves) interconnecting the sampling points of the input audio signal along the time axis, i.e. typically the x-axis representing the samples of the input audio signal, or the tangents representing in the respective zero-crossings of the respective graphical functions (curves) interconnecting the sampling points of the input audio signal along the time axis, i.e. typically the x-axis representing the samples of the input audio signal, in a clockwise or counterclockwise direction by predefinable or predefined degrees. Thus, the tilt function allows to consistently affect the acoustically perceptible properties of the modified audio signal.

Typically, the input audio signal that can be processed by or by the at least one audio processing device has a specific original waveform. The at least one audio processing device may be configured to modify a particular original waveform of the input audio signal into at least one target waveform of the modified audio signal. In particular, the at least one audio processing device may be configured to modify a particular original waveform of the audio signal based on the audio signal modification rule or the audio signal modification rule specifying a defined change in the waveform of the input audio signal from its original waveform to at least one target waveform of the modified audio signal. Thus, the at least one audio processing device may be configured to modify the original waveform of the input audio signal by applying at least one respective audio signal modification rule. Thus, modifying the original waveform of the input audio signal to or towards the target waveform of the modified audio signal allows to consistently affect the acoustically perceptible properties of the modified audio signal.

The respective target waveform of the input audio signal may be a symmetric waveform, in particular a rectangular waveform, a triangular waveform or a needle-like waveform. Alternatively, the respective target waveform of the input audio signal may be an asymmetric waveform, in particular a sawtooth waveform, preferably a straight or curved falling or rising sawtooth waveform. However, the corresponding target waveform may also be a free-form waveform.

The at least one audio processing device may be configured to: a skip rule or a skip factor is applied according to which at least one zero crossing between the first zero crossing and the further zero crossing is not taken into account for determining the interval between the first zero crossing and the further zero crossing of the audio signal. The application of the respective skip rule or the respective skip factor may allow to generate a modified audio signal having a very low frequency. As a general rule, the higher the skip factor, the lower the frequency of the modified audio signal. Thus, the application of the respective skip rule or the respective skip factor allows to consistently influence the acoustically perceptible property of the modified audio signal.

The apparatus may further comprise at least one filter means, in particular a low-pass filter means, arranged and/or configured to apply at least one filtering rule to the audio signal before the audio signal is processed by the audio processing means. The respective filter device is typically arranged on the input side of the audio processing device. Additionally or alternatively, the apparatus may further comprise at least one filter means, in particular a low pass filter means, arranged to apply at least one filtering rule to the audio signal after the audio signal has been processed by the audio processing means. The respective filter means are typically arranged on the output side of the audio processing means. The respective filter means typically positively influence the generation of respective harmonics of the low frequency components in the input audio signal, e.g. due to the ability to remove undesired intermodulation artifacts. The cut-off frequency of the respective filter means may be determined based on an operating parameter of the apparatus. For example only, the cut-off frequency of the respective filter means may be the lower-3 dB cut-off frequency of the device.

The apparatus may comprise one or more audio processing devices. The provision of a plurality of audio processing means allows to process more than one half-wave of the input audio signal at a time, thereby generating (sub-) harmonic low-frequency components.

Thus, the apparatus may comprise a first audio processing means and at least one further audio processing means.

The respective first audio processing device may be arranged in parallel with the respective at least one further audio processing device and vice versa. Thus, the apparatus may comprise a first audio processing device and at least one further audio processing device arranged in a parallel arrangement.

The respective first audio processing device may be configured to modify the input audio signal based on the first audio signal modification rule by: the positions of the sampling points in the first set of sampling points in the interval are changed such that each sampling point in the first set of sampling points changes from its respective first position in the first set of sampling points to its respective second position in the second set of sampling points. The respective at least one further audio processing device may be configured to: modifying the input audio signal based on at least one further audio signal modification rule by: the positions of the sampling points in the first set of sampling points in the interval are changed such that each sampling point in the first set of sampling points changes from its respective first position in the first set of sampling points to its respective second position in the second set of sampling points. Thus, the audio signal modification properties of the respective first audio processing device and the respective at least one further audio processing device may be at least partially different; typically, the likelihood of generating low frequency component harmonics of the input audio signal may be enhanced by using two or more audio processing devices.

Thus, a first audio signal modification rule of a respective first audio processing apparatus may specify a defined change of the waveform of the audio signal from its original waveform to at least one first target waveform of the audio signal, and at least another audio signal modification rule of a respective at least one further audio processing apparatus may specify a defined change of the waveform of the audio signal from its original waveform to at least another target waveform of the audio signal. Thereby, the first target waveform of the audio signal as specified by the at least one first audio signal modification rule may be opposite to the at least one further target waveform of the audio signal as specified by the at least one further audio signal modification rule. As an example of an asymmetric target waveform only, a first target waveform of the audio signal may be an ascending sawtooth waveform and at least one further target waveform of the audio signal may be a descending sawtooth waveform. Similar principles apply to other asymmetric waveforms. Similar principles apply to symmetric waveforms.

The respective first audio processing apparatus may be configured to apply a first skip rule or a first skip factor according to which at least one zero-crossing between the first zero-crossing and the further zero-crossing is not considered for determining an interval between the first zero-crossing and the further zero-crossing of the audio signal, and the respective at least one further audio processing apparatus may be configured to apply at least one further skip rule or at least one further skip factor according to which at least one zero-crossing between the first zero-crossing and the further zero-crossing is not considered for determining an interval between the first zero-crossing and the further zero-crossing of the audio signal. Thus, the first skip rule or the first skip factor, as applicable to the at least one first audio processing means, may be different from (i.e. higher or lower than) the at least one further skip rule or the at least one further skip factor, as applicable to the at least one further audio processing means. Applying different skip rules or skip factors accordingly allows to consistently influence the acoustically perceptible properties of the modified audio signal.

A second aspect of the invention relates to a device for outputting an audio signal, in particular in a vehicle cabin, comprising a plurality of samples, in particular in order to generate a missing low-frequency component in the audio signal. The apparatus comprises:

-at least one audio processing device configured to:

-processing an audio signal comprising a plurality of evenly spaced sampling points in a time-dependent representation of the audio signal, in particular in a half-wave representation of the audio signal;

-determining an interval between a first zero-crossing and a further zero-crossing of the audio signal;

-determining a first set of sample points in the interval, the first set of sample points comprising a plurality of sample points at a first position in the interval;

-determining a second set of sample points in the interval, the second set of sample points comprising a plurality of sample points at a second position in the interval;

-modifying the audio signal in the interval based on the audio signal modification rule by: changing the positions of the sampling points in the first set of sampling points in the interval such that each sampling point in the first set of sampling points changes from its respective first position in the first set of sampling points to its respective second position in the second set of sampling points;

-applying the modified audio signal interval to a corresponding interval of the original audio signal in order to generate a modified audio signal; and

at least one audio output device configured to output a modified audio signal in an acoustic environment, in particular in a vehicle cabin.

At least one audio output device includes one or more audio output elements, such as speakers. The at least one audio output element may be constructed as a specific bass audio output element, such as a woofer or a bass vibrator.

The device may particularly comprise audio processing means as specified by the device according to the first aspect of the invention.

All remarks made in relation to the device according to the first aspect of the invention are also applicable to the device of the second aspect of the invention.

A third aspect of the invention relates to a method for processing an audio signal comprising a plurality of samples, in particular in order to generate a missing low frequency component in the audio signal. The method comprises the following steps:

-processing an audio signal comprising a plurality of evenly spaced sampling points in a time-dependent representation of the audio signal, in particular in a half-wave representation of the audio signal;

-determining an interval between a first zero-crossing and a further zero-crossing of the audio signal;

-determining a first set of sample points in the interval, the first set of sample points comprising a plurality of sample points at a first position in the interval;

-determining a second set of sample points in the interval, the second set of sample points comprising a plurality of sample points at a second position in the interval;

-modifying the audio signal in the interval based on the audio signal modification rule by: changing the positions of the sampling points in the first set of sampling points in the interval such that each sampling point in the first set of sampling points changes from its respective first position in the first set of sampling points to its respective second position in the second set of sampling points;

-applying the modified audio signal interval to a corresponding interval of the original audio signal in order to generate a modified audio signal.

All comments made in respect of the device according to the first aspect of the invention also apply to the method of the third aspect of the invention.

A fourth aspect of the invention relates to a method for outputting an audio signal, in particular in a vehicle cabin, comprising a plurality of samples, in particular in order to generate a missing low-frequency component in the audio signal. The method comprises the following steps:

-processing an audio signal comprising a plurality of evenly spaced sampling points in a time-dependent representation of the audio signal, in particular in a half-wave representation of the audio signal;

-determining an interval between a first zero-crossing and a further zero-crossing of the audio signal;

-determining a first set of sample points in the interval, the first set of sample points comprising a plurality of sample points at a first position in the interval;

-determining a second set of sample points in the interval, the second set of sample points comprising a plurality of sample points at a second position in the interval;

-modifying the audio signal in the interval based on the audio signal modification rule by: changing the positions of the sampling points in the first set of sampling points in the interval such that each sampling point in the first set of sampling points changes from its respective first position in the first set of sampling points to its respective second position in the second set of sampling points;

-applying the modified audio signal interval to a corresponding interval of the original audio signal in order to generate a modified audio signal; and

-outputting the modified audio signal, in particular in a vehicle cabin.

All comments made in relation to the device according to the second aspect of the invention also apply to the method of the fourth aspect of the invention.

Drawings

Exemplary embodiments of the invention are described with reference to the accompanying drawings, whereby:

fig. 1 to 5 each show a schematic diagram of a device according to an exemplary embodiment;

FIG. 6 shows a time-dependent representation of an input audio signal prior to modification based on an audio signal modification rule according to an example embodiment;

FIG. 7 shows a time-dependent representation of an input audio signal after modification based on an audio signal modification rule according to an exemplary embodiment;

FIG. 8 shows a time-dependent representation of an input audio signal prior to modification based on an audio signal modification rule according to an example embodiment; and is

Fig. 9, 10 each show a time-dependent representation of an input audio signal after modification based on an audio signal modification rule according to an exemplary embodiment.

Detailed Description

Fig. 1 shows a schematic diagram of a device 1 for processing an audio signal comprising a plurality of samples according to an exemplary embodiment. The device 1 is particularly configured for processing an audio signal to generate (missing) harmonics of a low frequency component of the input audio signal.

The apparatus 1 comprises audio input means 2, i.e. means through which a digital input audio signal can be input to the apparatus 1, and audio output means 3, i.e. means through which a modified audio signal can be output in an acoustic environment. The audio input device 2 may include one or more audio input elements, such as a digital audio input interface. The audio output device 3 may comprise one or more audio output elements, such as a loudspeaker.

The apparatus 1 is generally applicable to any audio application in which a poor low frequency response is given, for example due to constructional and/or physical limitations of the audio output element (e.g. speaker). In other words, the apparatus 1 may be generally applied to any audio application in which virtual bass enhancement is used to compensate for missing low frequency component harmonics in an audio signal due to the construction and/or physical limitations of the audio output elements (e.g., speakers).

An exemplary audio application of the apparatus 1 is a mobile device application or a portable device application. Thus, the apparatus 1 may be installed in a mobile device or a portable device, such as a mobile computer, a smart phone, a tablet computer, a mobile speaker, etc.

Fig. 1 exemplarily shows a car audio application of the device 1. Thus, the device 1 may be installed in a vehicle 4 or car, respectively. The device 1 may accordingly be provided as a vehicle audio system or a car audio system, or the device 1 may accordingly form part of a vehicle audio system or a car audio system. In the automotive application of fig. 1, the apparatus 1 may allow compensation for low frequency component harmonics that are missing from an audio signal due to the configuration and/or physical limitations of an audio output element (e.g., speaker) disposed in the vehicle 4 or automobile, respectively.

In the exemplary embodiment of fig. 1, the device 1 comprises an audio processing means 5 embodied in hardware and/or software, an optional first filter means 6 connected to the audio processing means 5 at an input side of the audio processing means 5, an optional second filter means 7 connected to the audio processing means 5 at an output side of the audio processing means 5, an optional compensating delay means 8 arranged in parallel with the audio processing means 5, and an optional mixer means 9 connected to the second filter means at an output side of the second filter means 7 and to the delay means 8 at an output side of the delay means 8.

The audio processing means 5 are configured to process the input audio signal comprising a plurality of samples in a time-dependent representation of the input audio signal, in particular in a half-wave representation of the input audio signal. (see fig. 6). As can be seen from fig. 6, the time-dependent representation of the input audio signal is or comprises a time-dependent representation of spaced sampling points P1 of the input audio signal, more particularly a time-dependent representation of non-uniformly spaced sampling points P of the input audio signal. As can further be seen from fig. 6, the time-dependent representation of the input audio signal may comprise a graphical function (curve) interconnecting the sampling points P of the input audio signal along a time axis, i.e. an x-axis representing samples of the input audio signal. For example, the corresponding graphical function may be determined by interpolation of the sample points P of the input audio signal. Thus, the audio processing device 5 is configured to generate a time-dependent representation of the input audio signal, in particular a half-wave representation of the input audio signal, from the input audio signal comprising a plurality of samples. Thus, during operation of the device 1, the audio processing means 5 process the respective input audio signal in a time-dependent representation of the input audio signal, in particular in a half-wave representation of the input audio signal, and generate a time-dependent representation of the input audio signal, in particular a half-wave representation of the input audio signal, from the respective input audio signal.

The audio processing device 5 is further configured to determine an interval between a first zero-crossing and a further zero-crossing of the input audio signal in the time-dependent representation of the input audio signal. Thus, the audio processing device 5 is configured to analyze the zero-crossings of the time-dependent representation of the input audio signal, i.e. the positions at which the graphical function interconnecting the sampling points P of the input audio signal crosses the time axis in the time-dependent representation, and to determine, based on the determination of the respective zero-crossings, the interval between a first zero-crossing, a first position at which the graphical function interconnecting the sampling points P of the input audio signal crosses the time axis for the first time, and another zero-crossing (or a second zero-crossing), another position at which the graphical function interconnecting the sampling points P of the input audio signal crosses the time axis for another (or a second) time. Thus, during operation of the device 1, the audio processing means 5 analyze the respective zero-crossing of the time-dependent representation of the input audio signal and determine an interval I between the respective first zero-crossing and the respective further zero-crossing (or second zero-crossing) based on the determination of the respective zero-crossing.

The respective first zero-crossing and the further zero-crossing may be directly consecutive zero-crossings. However, it is also possible that the respective first zero-crossing and the further zero-crossing are not directly consecutive zero-crossings, but are indirectly consecutive zero-crossings, such that at least one zero-crossing is located between the respective first zero-crossing and the respective further zero-crossing. Thus, the respective interval I may extend between two directly consecutive zero-crossings of the time-dependent representation of the input audio signal, or the respective interval I may extend between two indirectly consecutive zero-crossings of the time-dependent representation of the input audio signal.

The audio processing apparatus 5 is further configured to determine a first set S1 of sample points P in the determined interval I, the first set of sample points P comprising a plurality of sample points P at a first position in the interval I (see fig. 6). Thus, during operation of the device 1, the audio processing means 5 determines a first set of S1 sample points P in the interval I, the first set of S1 sample points P comprising a plurality of sample points P at a first position in the interval I (see fig. 6). The positions of the sample points P in the first set S1 of sample points P in the interval I typically represent the original positions of the sample points P of the input audio signal in the interval I as given in the time-dependent representation of the input audio signal (see fig. 6). In other words, the positions of the sample points P of the first set S1 of sample points P typically correspond to the original positions of the sample points P of the input audio signal in the interval I as given in the time-dependent representation of the input audio signal obtained by processing the input audio signal.

The audio processing apparatus 5 is further configured to determine a second set of S2 sampling points in the determined interval I, the second set of S2 sampling points P comprising a plurality of sampling points P at a second position in the interval I (see fig. 7). Thus, during operation of the device 5, the audio processing apparatus 5 determines a second set of S2 sampling points in the interval I, the second set of S2 sampling points P including a plurality of sampling points P at second positions in the interval I (see fig. 7). The positions of the sample points P in the second set S2 of sample points P represent the target positions of the sample points P of the input audio signal in the interval I and are therefore offset from the original positions of the sample points P of the input audio signal in the interval I as given in the time-dependent representation of the input audio signal (see fig. 6, 7). In other words, the positions of the sample points P in the second set of S2 sample points P in the interval I typically correspond to the positions of the position offsets of the sample points P in the first set of S1 sample points P1 in the interval as given in the time-dependent representation of the input audio signal.

As can be seen from fig. 6, 7, the number of sampling points P in the first set of S1 sampling points P may be equal to the number of sampling points P in the second set of S2 sampling points P.

The audio processing means 5 are further configured to modify the audio signal in the interval I based on the audio signal modification rule by: the positions of the sample points P in the first set of S1 sample points P in the interval I are changed such that each sample point P in the first set of S1 sample points P changes from its respective first position in the first set of S1 sample points P (as indicated in fig. 6) to its respective second position in the second set of S2 sample points P (as indicated in fig. 7). Thus, during operation of the device 1, the audio processing means 5 change the positions of the sampling points P in the first set of S1 sampling points P in the interval I based on, i.e. using, the audio signal modification rule, and thus cause each sampling point P in the first set of S1 sampling points P to change from its respective first position (as indicated in fig. 6) in the first set of S1 sampling points P to its respective second position (as indicated in fig. 7) in the second set of S2 sampling points P. Accordingly, the audio signal modification rule may specify a change in the positions of the sample points P in the interval I such that the position of each sample point P changes from its initial position in the first set of S1 sample points (see fig. 6) to its target position in the second set of S2 sample points (see fig. 7). Accordingly, the modification rule may also specify an offset between the position of the respective sample point P in the first set of S1 sample points P (i.e., before the position of the respective sample point P has changed) and the changed position of the respective sample point P in the second set of S2 sample points P (i.e., after the position of the respective sample point P has changed).

The audio processing means 5 are further configured to apply the modified audio signal intervals to corresponding intervals of the original input audio signal in order to generate a modified audio signal. The application of the modified audio signal to the corresponding interval of the original input audio signal may also be performed by the mixer means 9. Thus, during operation of the device 1, the audio processing means 5 apply the modified audio signal intervals to the corresponding intervals of the original input audio signal in order to generate a modified audio signal. The modified audio signal is acoustically perceptible or acoustically perceptible as if the original input audio signal would include generated harmonics of the low frequency components. The modified audio signal is typically the input audio signal with a level that is constant such that no automatic gain control stage needs to be applied.

The modified audio signal may be output in an acoustic environment (e.g., a vehicle cabin) via audio output device 3.

Thus, as can be seen from the above description of the operation of the audio processing apparatus 5, the audio processing apparatus 5 is configured to resample an input audio signal having a plurality of samples, in particular on a non-uniformly spaced basis and in particular on a uniformly spaced basis, to again spread the samples by changing the position of a sample point P of the first set of S1 sample points P such that each sample point P of the first set of S1 sample points P changes from its respective first position in the first set of S1 sample points P to its respective second position in the second set of S2 sample points P.

As can be seen from the exemplary embodiments of fig. 6, 7, an input audio signal representing a positive pure sinusoidal half-wave may be resampled at a low sample point density at the beginning of the half-wave and at a higher and higher sample point density towards the end of the half-wave, which results in a waveform of the audio signal resembling a falling sawtooth waveform. As can be further seen in fig. 6 and 7, if the following negative half-wave is resampled at the inverse sample point density, the resulting audio signal will have the same fundamental frequency as the original sinusoidal half-wave but a harmonic pattern similar to a sawtooth half-wave.

The audio processing means 5 may be configured to determine the number of sample points P between the first zero-crossing and the at least one further zero-crossing such that it is the same as the number of sample points P in the corresponding interval I in the original input audio signal. The number of sample points P between the first zero crossing and the at least one further zero crossing is determined such that it is the same as the number of sample points P in the corresponding interval I in the original input audio signal, typically having a positive influence on the generation of low frequency component harmonics.

The audio processing apparatus 5 may be configured to modify the audio signal based on an audio signal modification rule specifying a definable or defined change of the positions of the sample points P in the first set of S1 sample points P in the interval I such that each sample point P in the first set of S1 sample points P changes from its respective first position in the first set of S1 sample points P (see fig. 6) to its respective second position S2 in the second set of S2 sample points P (see fig. 7).

As can be further seen from fig. 6, 7, the audio signal modification rule may particularly specify a defined variation of the positions of the sampling points P in the first set of S1 sampling points P in the interval I, such that each sampling point P in the first set of S1 sampling points P is changed from its respective first position in the first set of S1 sampling points P (see fig. 6) to its respective second position in the second set of S2 sampling points P (see fig. 7), such that the sampling points P in the second set of S2 sampling points P are equally or uniformly spaced. Accordingly, the audio processing device 5 may be configured to again equally or uniformly spread the samples by changing the positions of the sampling points P in the first set of S1 sampling points P on the premise that the positions of the sampling points P in the second set of S2 sampling points P are equally or uniformly spaced such that each sampling point P in the first set of S1 sampling points P changes from its respective first position in the first set of S1 sampling points P to its respective second position in the second set of S2 sampling points P.

The audio signal modification rule may be or may comprise a mapping function, in particular a monotonic mapping function, configured to map an input sample point P having a respective first position of the first set S1 sample points P1 to an output sample point P having a respective second position of the second set S2 sample points P. As can be seen from fig. 6, 7, the mapping function may specifically map input sample points P (see fig. 6) within a predefinable or predefined range (e.g. within a range of [0,1 ]) to output sample points P (see fig. 7) within a predefinable or predefined range. Thus, the audio processing device 5 may be configured to map the position of each sample point P of the first set S1 of sample points P to a defined position in the second set S2 of sample points P based on the respective mapping function. As can be seen from fig. 6, 7, the mapping function may specifically allow for evenly spaced positions of the sampling points P in the second set S2 of sampling points P.

Three examples of corresponding mapping functions f (x) are given below, where in brackets are the final waveform shape of the modified audio signal.

Example 1: (x) ═ e^x*D-1)/(e^D-1) (rising curved sawtooth wave form)

Example 2: (x) ═ e^D-e^xr*D)/(e^D-1) (descending curved sawtooth waveform)

Example 3: (x) log (1+ (x D))/log (1+ D) (descending straight sawtooth waveform)

Thus, x may be a function of the sampling points P in the second set of S2 sampling points P, whereby x (P) is P/(N-1), where N is the number of sampling points P in the second set of S2 sampling points P, where P is 0 for the first sampling point in the respective set and N-1 for the last sampling point P in the respective set. Thus, x (P) is in the range of [0,1 ].

The above exemplary mapping function f (x) is at [0,1]]Including predefinable or predefined distortion parameters D, and for the inverse input vector x_rIs operated in, wherein x_r(P)＝x(N-1)-x(P)。

Additionally or alternatively, the audio signal modification rule may be or may comprise a tilt function configured to tilt a zero-crossing tangent of the audio signal in a clockwise or counter-clockwise direction (see fig. 8 to 10). Thus, as indicated by the arrows in fig. 9, 10, the audio processing device 5 may be configured to tilt the zero-crossing tangent T of the input audio signal, i.e. the tangent in the respective zero-crossing of the respective graphical function (curve) interconnecting the sampling points P of the input audio signal along the time axis, i.e. typically the x-axis representing the samples of the input audio signal, in a clockwise direction (see fig. 9) or in a counter-clockwise direction (see fig. 10) by a predefinable or predefined number of degrees.

As can be seen for example from fig. 6 to 10, the input audio signal that can be processed by or by the audio processing device 5 has a specific original waveform. As can be seen from the above description in the context of fig. 6 to 10, the audio processing apparatus 5 is configured to modify a specific original waveform of an input audio signal into at least one target waveform of a modified audio signal. In particular, the audio processing apparatus 5 may be configured to modify a particular original waveform of the audio signal based on an audio signal modification rule specifying a defined change of the waveform of the input audio signal from its original waveform to at least one target waveform of the modified audio signal. Thus, the audio processing apparatus 5 may be configured to modify the original waveform of the input audio signal by applying at least one respective audio signal modification rule.

The corresponding target waveform of the input audio signal may be a symmetric waveform (see fig. 10), in particular a rectangular waveform, a triangular waveform or a needle-like waveform. Alternatively, the respective target waveform of the input audio signal may be an asymmetric waveform (see fig. 9), in particular a sawtooth waveform, preferably a straight or curved falling or rising sawtooth waveform. However, the corresponding target waveform may also be a free-form waveform.

The audio processing means 5 are configured to apply a skip rule or a skip factor according to which the interval I between a first zero crossing and a further zero crossing of the audio signal is determined without taking into account at least one zero crossing between the first zero crossing and the further zero crossing. The application of the respective skip rule or the respective skip factor may allow to generate a modified audio signal having a very low frequency. As a general rule, the higher the skip factor, the lower the frequency of the modified audio signal.

In the exemplary embodiment of fig. 1, the optional first filter means 6 is embodied as a low-pass filter, for example a low-pass filter with a cut-off frequency of 100Hz, and the optional second filter means 7 is embodied as a second low-pass filter, for example a low-pass filter with a cut-off frequency of 1000 Hz. However, other cut-off frequencies are contemplated.

Fig. 2 shows a schematic diagram of a device 1 according to another exemplary embodiment. The exemplary embodiment of the device of fig. 2 differs from the preceding embodiments in an optional further filter arrangement 10 connected to the delay arrangement 8 at the input side of the delay arrangement 8. The further filter arrangement 10 may be embodied as a parametric EQ filter. The further filter device 10 may have a center frequency of 160 Hz. However, other center frequencies are contemplated.

The exemplary embodiments of fig. 3 to 5 each show an apparatus 1 comprising a plurality of audio processing means 5 allowing to process more than one half-wave of an input audio signal at a time, thereby generating (sub-) harmonic low-frequency components.

As can be seen from the embodiments of fig. 3 to 5, the respective audio processing devices 5 may be arranged in parallel.

Fig. 3 shows a schematic diagram of a device 1 comprising a plurality of audio processing means 5 according to an exemplary embodiment. In this exemplary embodiment, the first audio processing device 5.1 (upper audio processing device 5) is configured to implement an audio signal modification rule that modifies an original waveform of an input audio signal into at least one first target waveform of a modified audio signal, and the second audio processing device 5.2 (lower audio processing device 5) is configured to implement an audio signal modification rule that modifies an original waveform of an input audio signal into at least one second target waveform of a modified audio signal. For example, the first target waveform may be a straight-rising sawtooth waveform. For example, the second target waveform may be a straight-down straight sawtooth waveform.

Thus, fig. 3 shows that the first audio processing device 5.1 may be configured to modify the input audio signal by changing the position of a sample point P of the first set of S1 sample points P in the interval I, based on the first audio signal modification rule, such that each sample point P of the first set of S1 sample points P changes from its respective first position in the first set of S1 sample points P to its respective second position in the second set of S2 sample points P. The respective other audio processing device 5.2 may be configured to modify the input audio signal by changing the position of a sample point P of the first set of S1 sample points P in the interval, based on at least one other audio signal modification rule, such that each sample point P of the first set of sample points P changes from its respective first position in the first set of S1 sample points P to its respective second position in the second set of S2 sample points P. Thus, the audio signal modification properties of the first audio processing device 5.1 and the further audio processing device 5.2 may differ at least partly.

Thus, a first audio signal modification rule of a respective first audio processing apparatus may specify a defined change of the waveform of the audio signal from its original waveform to at least one first target waveform of the audio signal, and at least another audio signal modification rule of a respective at least another audio processing apparatus may specify a defined change of the waveform of the audio signal from its original waveform to at least another target waveform of the audio signal. Thus, the first target waveform of the audio signal as specified by the at least one first audio signal modification rule may be opposite to the at least one further target waveform of the audio signal as specified by the at least one further audio signal modification rule.

In addition, the first audio processing apparatus 5.1 may be further configured to apply a first skip rule or a first skip factor according to which the interval I between the first zero-crossing and the further zero-crossing of the audio signal is determined without taking into account at least one zero-crossing between the first zero-crossing and the further zero-crossing, and the further audio processing apparatus 5.2 may be configured to apply at least one further skip rule or at least one further skip factor according to which the interval I between the first zero-crossing and the further zero-crossing of the audio signal is determined without taking into account at least one zero-crossing between the first zero-crossing and the further zero-crossing. Thus, the first skip rule or first skip factor as applicable to the first audio processing 5.1 device may be equal to or different (i.e. higher or lower) than the further skip rule or further skip factor as applicable to the further audio processing device 5.2. In the exemplary embodiment of fig. 3, the skip factors of the audio processing means 5.1, 5.2 are equal. However, different skip factors are contemplated.

Fig. 3 also shows a first optional filter 6.1 connected to the first audio processing means 5.1 at the input side of the first audio processing means 5.1 and a second optional filter 6.2 connected to the second audio processing means 5.2 at the input side of the second audio processing means 5.2. The optional filter means 6.1, 6.2 may be embodied as a low-pass filter. The optional filter means 6.1, 6.2 may have the same or different cut-off frequencies. For example, the first filter 6.1 may have a cut-off frequency of 100Hz and the second filter 6.2 may have a cut-off frequency of 50 Hz. However, other cut-off frequencies are contemplated.

Fig. 3 also shows an optional further filter 7 connected to the first mixer arrangement 9.1 at the output side of the first mixer arrangement 9.1. An optional further filter means 7 may be embodied as a low-pass filter. An optional further filter arrangement 7 may have a cut-off frequency of 1000 Hz. However, other cut-off frequencies are contemplated.

Fig. 3 also shows an optional further mixer arrangement 9.2 connected to the further filter arrangement 7 at the output side of the further filter arrangement 7.

Fig. 4 shows a schematic diagram of a device 1 according to another exemplary embodiment. The exemplary embodiment of the apparatus of fig. 4 differs from the previous embodiments in that an additional audio output means 3.2, for example embodied as a bass vibrator, is connected to the optional filter means 7 at the output side of the optional filter means 7.

In the exemplary embodiment of fig. 4, the optional further filter means 7 may be embodied as a low-pass filter. The further filter means 7 may have a cut-off frequency of 25 Hz. However, other cut-off frequencies are contemplated.

Fig. 5 shows a schematic diagram of a device 1 according to another exemplary embodiment. The embodiment of fig. 5 generally indicates that the apparatus 1 may comprise a plurality of audio processing devices 5, a plurality of filter devices (indicated by the boxes representing filter banks) connected at the input side of the respective audio processing devices 5, and a plurality of filter devices (indicated by the boxes representing filter arrays) connected at the output side.

Each device 1 according to the embodiment of the figure generally allows implementing a method of processing an audio signal, comprising the steps of:

-processing an audio signal comprising a plurality of evenly spaced sampling points P in a time-dependent representation of the audio signal, in particular in a half-wave representation of the audio signal;

-determining an interval I between a first zero-crossing and a further zero-crossing of the audio signal;

-determining a first set of S1 sample points P in the interval, the first set of S1 sample points P comprising a plurality of sample points P at a first position in the interval I;

-determining a second set of S2 sample points P in the interval, the second set of S2 sample points P comprising a plurality of sample points P at a second position in the interval I;

-modifying the audio signal in interval I based on the audio signal modification rule by: changing the positions of the sample points P in the first set of S1 sample points P in the interval I such that each sample point P in the first set of S1 sample points P changes from its respective first position in the first set of S1 sample points P to its respective second position in the second set of S2 sample points P;

-applying the modified audio signal interval to a corresponding interval of the original audio signal in order to generate a modified audio signal.

Each device 1 according to the embodiment of the figure generally allows implementing a method for outputting audio signals, in particular in the vehicle cabin, comprising the steps of:

-processing an audio signal comprising a plurality of evenly spaced sampling points in a time-dependent representation of the audio signal, in particular in a half-wave representation of the audio signal;

-determining an interval I between a first zero-crossing and a further zero-crossing of the audio signal;

-determining a first set of S1 sample points P in interval I, the first set of S1 sample points P comprising a plurality of sample points P at a first position in interval I;

-determining a second set of S2 sample points P in interval I, the second set of S2 sample points P comprising a plurality of sample points P at a second position in interval I;

-modifying the audio signal in interval I based on the audio signal modification rule by: changing the positions of the sample points P in the first set of sample points P in the interval I such that each sample point P in the first set of S1 sample points P changes from its respective first position in the first set of S1 sample points P to its respective second position in the S2 second set of sample points P;

-applying the modified audio signal interval to the corresponding interval I of the original audio signal in order to generate a modified audio signal;

-outputting the modified audio signal, in particular in a vehicle cabin.

One or more particular features of the first exemplary embodiment may be combined with one or more particular features of at least one other exemplary embodiment.