HK1069705B - Method and device for controlling the bass reproduction of audio signals in electroacoustic transducers - Google Patents

Description

Method and apparatus for controlling bass reproduction of audio signal in electroacoustic transducer

Technical Field

The present invention relates to a method for controlling the bass reproduction of an audio signal in an electroacoustic transducer, and to an apparatus for controlling the bass reproduction of an audio signal in an electroacoustic transducer.

Background

The bass reproduction of audio signals in electroacoustic transducers, in particular in loudspeakers or headphones, is determined by the size of the electroacoustic transducers, the loudspeakers and the headphones. The smaller the loudspeaker diaphragm and its maximum excursion, the higher the resonance frequency at that.

A typical frequency curve for a small loudspeaker is shown in fig. 1. Electro-acoustic devices that employ such small electro-acoustic transducers and thus the bass reproduction described is unsatisfactory are mainly found in audio devices (devices for outputting or reproducing audio signals) of communication technology and information technology as well as entertainment electronics and consumer electronics, such as wireless mobile and cordless telephone handsets, notebooks, personal digital assistants, pocket radios, wireless alarm clocks, portable music playback devices, and the like.

In order to improve the bass reproduction with small loudspeakers, a known psychoacoustic principle can be used. This principle is called "Residual Hearing of Missing fundamental" or "Virtual Pitch".

The perception of the fundamental frequency can be simulated by harmonic combination according to this principle. It is thus also possible to simulate the perception of low frequencies by means of corresponding combinations of their harmonics.

For a detailed description of the "Virtual Pitch" principle, see e.zwicker and h.fastl publication "psychoacoustic", Springer publication, second edition 1999.

Methods based on the psychoacoustic principle are disclosed in US 6,111,960 and US 5,930,373, which generate a series of corresponding harmonics by means of the audio signal to simulate frequencies below a limit frequency.

A method based on the psycho-acoustic principle is disclosed in WO 00/15003, in which resonant frequencies already present in the audio signal are amplified. In order to improve the low-frequency reproduction of an audio signal in an electroacoustic transducer, the low-frequency components of the audio signal are isolated into a low-frequency audio signal, the isolated low-frequency components are filtered with a plurality of band-pass filters, the band-pass filtered frequency components are amplified in an amplifier with a controllable amplification factor, wherein the amplification factor is obtained from the envelope of the band-pass filtered frequency components, and a simulated low-frequency audio signal is generated by combining the original audio signal with the amplified frequency components.

Disclosure of Invention

The object on which the invention is based is to control the bass reproduction of an audio signal in an electroacoustic transducer on the basis of the described psychoacoustic principle, which is referred to as "Virtual Pitch" or "listening bass", in such a way that the perception of Virtual bass reproduction of the audio signal is improved with respect to the state of the art.

This object is solved not only by the method according to the invention but also by the device according to the invention.

The method according to the invention for controlling the bass reproduction of an audio signal in an electroacoustic transducer, wherein, a) the frequency components of the audio signal are isolated, and amplified with an amplification factor calculated from the audio signal, b) the frequency components of the amplified audio signal and the audio signal are combined in this way, so that a modified audio signal is generated, c) said modified audio signal is transmitted to said electroacoustic transducer, characterized in that d) the audio signal is band-pass filtered for isolating and amplifying first frequency components, e) for calculating the amplification factor, e1) the audio signal is low-pass filtered and/or band-pass filtered for isolating second frequency components, e2) the envelope and/or energy of the unfiltered, low-pass filtered and/or band-pass filtered audio signal is calculated.

The device according to the invention for controlling the reproduction of bass sounds in an electroacoustic transducer, wherein (a) there are isolating means to the input of which the audio signal is applied and which are used to isolate the frequency components of the audio signal, (b) there are calculating means to calculate an amplification factor from the audio signal, (c) there is an amplifier which is connected to the isolating means and to the calculating means in such a way that the frequency components of the audio signal are amplified with the calculated amplification factor, (d) there are combining means to which the audio signal and the frequency components of the amplified audio signal are applied to the input of the combining means and which combine the audio signal and the amplified frequency components of the audio signal in such a way that there is a modified audio signal at the output of the combining means which is adapted to the electroacoustic transducer, characterized in that (e) at least one band-pass filter or at least one band-pass filter and a low-pass filter, respectively, is provided for isolating a first frequency component and a second frequency component of the audio signal, (f) one of the band-pass filters is connected on the output side to the amplifier for isolating the first frequency component, (g) means are provided for calculating a signal envelope and/or a signal energy, wherein an unfiltered, low-pass filtered and/or band-pass filtered audio signal is applied to its input side, (h) the calculating means for calculating an amplification factor are connected on the input side to the means for calculating a signal envelope and/or a signal energy and on the output side to the amplifier for adjusting the amplification factor.

The idea behind the invention is that the reproduction of low frequencies or bass sounds output in said electroacoustic transducer is controlled in a simulated sense by the amplification of harmonics contained in the audio signal, so that the listener receives or perceives an improved bass reproduction. The control or simulation can be implemented both digitally by a program module in the digital signal processor DSP of the electroacoustic device outputting and/or reproducing the audio signal with the electroacoustic transducer, and analogically by a hardware circuit between the digital/analog transducer and the final amplifier of the electroacoustic device outputting and/or reproducing the audio signal with the electroacoustic transducer.

By means of the program module and the hardware circuit, only harmonics above the resonance frequency of the electroacoustic transducer, in particular of the loudspeaker, are amplified to simulate the perception of a fundamental frequency. The harmonic extraction or isolation is implemented in the program module by means of bandpass filtering, the amplification of the harmonic is implemented in the hardware circuit by means of a bandpass filter, and the amplification of the harmonic is Controlled by the amplification factor in the program module on the basis of software, and the amplification factor is Controlled in the hardware circuit by means of a Gain control Amplifier (Gain Controlled Amplifier) designed for the harmonic. The amplification factor is preferably controlled by the frequency content of the audio signal below the resonance frequency or limit frequency of the electroacoustic transducer.

The method according to the invention has the advantage that the amplification of the original harmonics of the resonances present in the audio signal ensures a significantly improved quality of the modified audio signal generated in the digital signal processor. Distortion of the audio signal is thereby avoided, among other things. In addition, the method of the invention has smaller requirements on the computing power and the memory capacity in the digital signal processor.

The invention also comprises advantageous developments based on the above-described method and device.

If the audio signal combined with the amplified frequency components is buffered in case a "finite impulse response" filter is used-AS opposed to an "infinite impulse response" filter-in order to compensate for the phase offset between the amplified frequency components and the audio signal resulting from the use of a FIR filter for the combination, it is advantageous if the band-pass filtering is implemented by means of a "finite impulse response" filter (FIR-F) or if the Audio Signal (AS) combined with the amplified frequency components (VFK) is buffered (ZWS).

If the modified Audio Signal (AS) is filtered to amplify a selected frequency in order to improve the quality of the modified audio signal output by the electroacoustic transducer, it is advantageous if the modified Audio Signal (AS) is filtered (PRF, PRF1) in order to amplify the selected frequency or if the method is implemented in an electronic device for outputting and/or reproducing audio signals.

Drawings

Two embodiments of the invention are explained with the aid of fig. 2 to 7. Wherein:

figure 2 shows a digital implementation of the inventive method implemented in the form of program modules in a digital signal processor of an electronic wireless device for outputting and/or playing back audio signals,

figure 3 shows an analog implementation of the inventive arrangement in a hardware solution of an electronic radio device for outputting and/or playing back audio signals,

figure 4 shows a first embodiment of a program module according to figure 2,

figure 5 shows a second embodiment of a program module according to figure 2,

figure 6 shows a third embodiment of a program module according to figure 2,

fig. 7 shows an embodiment of the control device according to fig. 3.

Detailed Description

Fig. 2 shows, as a second exemplary embodiment, a speech processing circuit for outputting and/or reproducing audio signals, in particular speech signals, in a wireless device FG in the form of a circuit diagram or a circuit block diagram, wherein the present invention is implemented (digitally implemented) in a program module PGM of a digital signal processor DSP. The radio device FG receives an analog radio signal FS, over which the encoded speech information is modulated, via an antenna ANT. In the receiver EMP, a digital demodulation signal DDS is generated from the analog radio signal FS, which is modulated and encoded, by means of a microprocessor MP and an analog-to-digital converter ADW. The digital demodulation signal DDS is then transmitted to the speech decoder SDK of the digital signal processor DSP. In the speech decoder SDK, a speech signal or, in general, an audio signal AS, is generated from the digital demodulation signal DDS. The audio signal AS is then transmitted to the program module for controlling the bass reproduction of the audio signal in the electroacoustic transducer PGM of the digital signal processor DSP. A modified audio signal MAS is generated from the audio signal AS in the program module PGM of the digital signal processor DSP, which signal is then filtered by the filter FIL of the digital signal processor DSP. The filtered modified audio signal MAS is finally transmitted to a digital-to-analog converter DAW and then amplified in a final amplifier EVS, after which the speech information contained in the modified audio signal MAS is output by an electroacoustic converter EAW, which is preferably designed as a loudspeaker.

Fig. 3 shows as a second embodiment a speech processing circuit in a wireless device FG in the form of a circuit diagram or a circuit block diagram, in which, in contrast to fig. 2, the invention is implemented in an electroacoustic transducer STV in an apparatus for controlling the bass reproduction of audio signals, outside a speech signal processor DSP in the analog part of the wireless device FG (analog implementation). Speech processing is started in the wireless device FG, which starts again with the transmission of the analog radio signal FS, on which encoded speech information has been modulated, to the receiver EMP via the antenna ANT. The digital demodulation signal DDS is again generated in the receiver EMP from the analog radio signal FS by means of the microprocessor MP and the analog-to-digital converter ADW. The digital demodulation signal DDS is then transmitted again to the speech decoder SDK in the digital signal processor SDP. The decoded speech signal or the decoded audio signal AS in its entirety in the general sense is again obtained from the digital demodulation signal DDS in the speech decoder SDK. The audio signal AS is subsequently filtered in the filter FIL of the digital signal processor DSP, after which the filtered audio signal is correspondingly transformed in the digital-to-analog converter DAW. The converted audio signal AS is subsequently transmitted to the means for controlling the bass reproduction of the audio signal in the electroacoustic transducer STV, where a modified audio signal MAS is generated from the audio signal AS. The modified audio signal MAS is subsequently amplified in the final amplifier EVS, after which the speech information contained in the modified audio signal MAS is output by an electroacoustic transducer EAW, which is again preferably designed as a loudspeaker.

Fig. 4 shows a first embodiment of a program module according to fig. 2. For isolating the first frequency component FK, the audio signal AS is bandpass filtered by means of a bandpass filter BPF implemented in software, and for isolating the second frequency component FK' is lowpass filtered by means of a lowpass filter TPF implemented in software. When the first frequency component FK is amplified, an amplification factor VF that determines the gain of the first frequency component FK is generated using the second frequency component FK'.

Alternatively, the low-pass filter TPF can be replaced by a further bandpass filter implemented by software or even a bandpass filter BPF generating the first frequency component FK. In the latter case, the two frequency components FK, FK 'are identical (FK ═ FK').

The band-pass filter BPF is preferably designed as a "finite impulse response" filter (FIR filter) FIR-F or as an "infinite impulse response" filter (IIR filter) IIR-F. If the band-pass filter BPF is designed AS a "finite impulse response" filter FIR-F, the program module PGM contains an intermediate memory ZWS for buffering the audio signal AS. If the band-pass filter BPF is constructed as an "infinite impulse response" filter IIR-F, the intermediate memory ZWS is not required. To illustrate this in fig. 4, the intermediate memory ZWS is depicted as a dashed block.

The band-pass filtered audio signal FK and the frequency components FK isolated by the band-pass filter BPF are output for amplification to an input of an amplifier VS controlled by an amplification factor VF and implemented by software. In order to obtain the amplification factor VF, means MBSE implemented by software for calculating the signal envelope and/or the signal energy are provided in the program module PGM, which means supply an input value from the low-pass filtered audio signal FK' to a means MBVF also implemented by software, which latter means MBVF is used for calculating the amplification factor of the program module PGM. The computation means MBVF then provide an amplification factor VF by means of which the amplifier VS can be controlled. At the output of the amplifier VS, a bandpass-filtered audio signal VSFK is provided, which is amplified by an amplification coefficient VF. The amplified bandpass-filtered audio signal VSFK and the optionally intermediately stored audio signal AS are then combined or added by means of the combining means KM, which is preferably designed AS adding means and implemented by means of software, of the program module PGM. As a result of these operations, a modified audio signal MAS is generated which, for improving the signal quality, is preferably filtered by means of a representation filter (Pr filter) PRF implemented by means of software. However, it is also possible to transmit the modified audio signal MAS to the filter FIL without further filtering by means of the representation filter PRF, as explained in the description of fig. 2.

Fig. 5 shows a second embodiment of the program module PGM according to fig. 2 according to fig. 4. The audio signal AS is again band-pass filtered by means of the band-pass filter BPF for isolating the first frequency component FK and low-pass filtered by means of the low-pass filter TPF for isolating the second frequency component FK'. When the first frequency component FK is amplified again, an amplification factor VF that determines the gain of the first frequency component FK is generated again using the second frequency component FK'.

Again, another band-pass filter or even a band-pass filter BPF generating the first frequency component FK can optionally be used instead of the low-pass filter TPF. In the latter case, the two frequency components FK, FK 'are once again identical (FK ═ FK').

The band-pass filter BPF is again preferably designed as a "finite impulse response" filter (FIR filter) FIR-F or as an "infinite impulse response" filter (IIR filter) IIR-F. If the band-pass filter BPF is constructed AS a "finite impulse response" filter (FIR filter) FIR-F, the program module PGM again contains the intermediate memory ZWS for buffering the audio signal AS. If the band-pass filter BPF is designed as an "infinite impulse response" filter (IIR filter) IIR-F, the intermediate memory ZWS is again not required. To illustrate this in fig. 5, the intermediate memory ZWS is depicted as a dashed block.

The band-pass filtered audio signal FK and the frequency components FK isolated by the band-pass filter BPF are output for amplification as in fig. 4 to an input of an amplifier VS controllable by an amplification factor VF. In order to obtain the amplification factor VF, means MBSE for calculating the signal envelope and/or the signal energy are again provided in the program module PGM, which means supply an input value again from the low-pass filtered audio signal FK' to the means MBVF for calculating the amplification factor of the program module PGM.

In the embodiment of the program module according to fig. 5, in contrast to the embodiment according to fig. 4, a further input value is supplied to the computation means MBVF, wherein the input value is determined by a further means MBSE for computing the signal envelope and/or the signal energy. Said further input value is calculated by said calculating means MBSE from said unfiltered audio signal AS.

The computation means MBVF then supplies the amplification factor VF, which again enables the amplifier VS to be controlled, from the two input values. At the output of the amplifier VS, a bandpass-filtered audio signal VSFK amplified by an amplification factor VF is thus again provided. The amplified bandpass-filtered audio signal VSFK and the optionally intermediately stored audio signal AS are then combined or added again by means of the combining means KM of the program module PGM, which is again preferably designed AS adding means. As a result of these operations, the modified audio signal MAS is generated which, for the purpose of improving the signal quality, is preferably filtered again by means of the representation filter PRF. However, it is also possible to transmit the modified audio signal MAS again to the filter FIL without further filtering by means of the representation filter PRF, as explained in the description of fig. 2.

Fig. 6 shows a third embodiment of the program module PGM according to fig. 2 according to fig. 4. The audio signal AS is once again band-pass filtered by means of the band-pass filter BPF in order to isolate the first frequency component FK and once again low-pass filtered by means of the low-pass filter TPF in order to isolate the second frequency component FK'. When the first frequency component FK is amplified again, an amplification factor VF that determines the gain of the first frequency component FK is generated again using the second frequency component FK'.

Again, another band-pass filter or even a band-pass filter BPF generating the first frequency component FK can optionally be used instead of the low-pass filter TPF. In the latter case, the two frequency components FK, FK 'are identical (FK ═ FK').

The band-pass filter BPF is again preferably designed as a "finite impulse response" filter (FIR filter) FIR-F or as an "infinite impulse response" filter (IIR filter) IIR-F. If the band-pass filter BPF is constructed AS a "finite impulse response" filter (FIR filter) FIR-F, the program module PGM again contains the intermediate memory ZWS for buffering the audio signal AS. If the band-pass filter BPF is designed as an "infinite impulse response" filter (IIR filter) IIR-F, the intermediate memory ZWS is again not required. To illustrate this in fig. 6, the intermediate memory ZWS is depicted as a dashed block.

The band-pass filtered audio signal FK and the frequency components FK isolated by the band-pass filter BPF are output for amplification as in fig. 4 and 5 to an input of an amplifier VS controllable by an amplification factor VF. In order to obtain the amplification factor VF, means MBSE for calculating the signal envelope and/or the signal energy are again provided in the program module PGM, which means supply an input value from the low-pass filtered audio signal FK' to the means MBVF for calculating the amplification factor of the program module PGM.

In the embodiment of the program module according to fig. 6, in contrast to the embodiment according to fig. 4, a further input value is supplied to the computation means MBVF, wherein the input value is determined by a further means MBSE for computing the signal envelope and/or the signal energy. Said further input value, which differs from the input value according to fig. 5, is calculated by said calculation means MBSE from said band-pass filtered audio signal AS.

The computation means MBVF then supplies the amplification factor VF, by means of which the amplifier VS can be controlled, from these two input values. At the output of the amplifier VS, a bandpass-filtered audio signal VSFK amplified by an amplification factor VF is thus again provided. The amplified bandpass-filtered audio signal VSFK and the optionally intermediately stored audio signal AS are then combined or added again by means of the combining means KM of the program module PGM, which is preferably designed AS an adding means. As a result of these operations, the modified audio signal MAS is generated again, which is preferably filtered again by the representation filter PRF for the purpose of improving the signal quality. However, it is also possible to transmit the modified audio signal MAS again to the filter FIL without further filtering by means of the representation filter PRF, as explained in the description of fig. 2.

Fig. 7 shows an embodiment of the control device STV according to fig. 3. The audio signal AS is band-pass filtered by means of a band-pass filter BPF1, which is designed AS a hardware module, for isolating the first frequency component FK, and low-pass filtered by means of a low-pass filter TPF1, which is designed AS a hardware module, for isolating the second frequency component FK'. When the first frequency component FK is amplified, an amplification factor VF that determines the gain of the first frequency component FK is generated using the second frequency component FK'.

Alternatively, the low-pass filter TPF1 can be replaced by a further band-pass filter designed as a hardware module or even a band-pass filter BPF1 that generates the first frequency component FK. In the latter case, the two frequency components FK, FK 'are identical (FK ═ FK').

The band-pass filtered audio signal FK and the frequency components FK isolated by means of the band-pass filter BPF1 are output for amplification to an input of an amplifier VS1, which is designed as a hardware module and is controllable by an amplification factor VF. In order to obtain the amplification VF, the control device STV has a device MBSE1 designed as a hardware module for calculating the signal envelope and/or the signal energy, wherein the device is preferably formed by a rectifier GLR and a further low-pass filter TPF2 connected in series and provides an input value from the low-pass filtered audio signal FK' to a device MBVF1 also designed as a hardware module, wherein the latter device MBVF1 is used for calculating the amplification of the control device STV. The computation means MBVF1 then supplies an amplification factor VF by means of which the amplifier VS1 can be controlled. At the output of the amplifier VS1, a bandpass-filtered audio signal VSFK amplified by an amplification factor VF is provided. The amplified bandpass-filtered audio signal VSFK and the audio signal AS are then combined or added by means of a combining means KM1, which is preferably designed AS an adding means and AS a hardware module, and which controls the device STV. As a result of these operations, the modified audio signal MAS is generated which, for the purpose of improving the signal quality, is preferably filtered by a representation filter PRF1 which is designed as a hardware module. However, it is also possible to transmit the modified audio signal MAS, as explained in the description of fig. 3, to the final amplifier EVS without further filtering by means of the representation filter PRF.

Claims (13)

1. Method for controlling the bass reproduction of an audio signal in an electroacoustic transducer, in which

a) The frequency components (FK, FK') of the Audio Signal (AS) are isolated and amplified (VS, VS1) by an amplification factor (VF) calculated from the Audio Signal (AS),

b) the frequency components (VSFK) of the amplified Audio Signal (AS) and the Audio Signal (AS) are combined (KM, KM') in such a way that a Modified Audio Signal (MAS) is generated,

c) the Modified Audio Signal (MAS) is transmitted to the electroacoustic transducer (EAW),

it is characterized in that the preparation method is characterized in that,

d) the Audio Signal (AS) is band-pass filtered (BPF, BPF1) for isolating and amplifying the first frequency component (FK),

e) for calculating (MBVF, MBVF1) the amplification factor (VF),

e1) the Audio Signal (AS) is low-pass filtered and/or band-pass filtered (BPF, BPF1, TPF1) to isolate a second frequency component (FK'),

e2) the envelope and/or the energy of the unfiltered, low-pass filtered and/or band-pass filtered audio signal (AS, FK') is calculated (MBSE, MBSE 1).

2. The method according to claim 1,

the band-pass filtering is implemented by means of a "finite impulse response" filter (FIR-F).

3. The method according to claim 1,

the band-pass filtering is implemented by means of an "infinite impulse response" filter (IIR-F).

4. The method according to claim 2,

the Audio Signal (AS) combined with the amplified frequency components (VFK) is buffered (ZWS).

5. Method according to one of claims 1 to 4,

for isolating and amplifying the frequency components and for calculating the amplification factor, band-pass filtering is carried out by means of a separate band-pass filter (BPF, BPF 1).

6. Method according to one of claims 1 to 4,

in order to isolate and amplify the frequency components, a bandpass filter (BPF, BPF1) is used for bandpass filtering, and in order to calculate the amplification factor, another bandpass filter is used for bandpass filtering.

7. The method according to claim 1,

the modified Audio Signal (AS) is filtered (PRF, PRF1) for amplifying the selected frequencies.

8. The method according to one of claims 1 to 4 or 7,

the method is implemented in an electronic device for outputting and/or reproducing audio signals.

9. The method according to claim 5,

the method is implemented in an electronic device for outputting and/or reproducing audio signals.

10. The method according to claim 6,

the method is implemented in an electronic device for outputting and/or reproducing audio signals.

11. Apparatus for controlling the reproduction of bass sounds in an electroacoustic transducer, in which

(a) Having an isolation device (BPF, BPF1, TPF1), to the input of which the Audio Signal (AS) is applied and which serves to isolate the frequency components (FK, FK'),

(b) having calculation means (MBVF, MBVF1) for calculating an amplification factor (VF) from the Audio Signal (AS),

(c) having an amplifier (VS, VS1) which is connected to the isolation device and the computation device in such a way that the frequency components (FK, FK') of the Audio Signal (AS) are amplified by means of the computed amplification factor (VF),

(d) having a combination device (KM, KM1), wherein the Audio Signal (AS) and the frequency components (VSFK) of the amplified Audio Signal (AS) are applied to an input of the combination device, and wherein the combination device (KM, KM1) combines the Audio Signal (AS) and the amplified frequency components (VSFK) of the Audio Signal (AS) in such a way that at an output of the combination device (KM, KM1) there is a Modified Audio Signal (MAS) which is suitable for the electroacoustic transducer (EAW),

it is characterized in that the preparation method is characterized in that,

(e) having at least one band-pass filter (BPF, BPF1), or respectively having at least one band-pass filter (BPF, BPF1) and a low-pass filter (TPF, TPF1), for isolating a first frequency component (FK) and a second frequency component (FK') of the Audio Signal (AS),

(f) one of the band-pass filters (BPF, BPF1) is connected on the output side to the amplifier (VS, VS1) for isolating the first frequency component (FK),

(g) having means for calculating a signal envelope and/or a signal energy (MBSE, MBSE1), wherein an unfiltered, low-pass filtered and/or band-pass filtered audio signal (AS, FK') is applied to its input side,

(h) the computation device (MBVF, MBVF1) for computing the amplification factor (VF) is connected on the input side to the device for computing the signal envelope and/or the signal energy (MBSE, MBSE1) and on the output side to the amplifier (VS, VS1) for setting the amplification factor (VF).

12. The apparatus of claim 11,

there is a representation filter (PRF, PRF1) for amplifying selected frequencies of the Modified Audio Signal (MAS).

13. The device according to claim 11 or 12,

the apparatus is integrated or comprised in an electronic device for outputting and/or reproducing audio signals.