EP1428411B1 - Method and device for controlling the bass reproduction of audio signals in electroacoustic transducers - Google Patents

Method and device for controlling the bass reproduction of audio signals in electroacoustic transducers Download PDF

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Publication number
EP1428411B1
EP1428411B1 EP01980187A EP01980187A EP1428411B1 EP 1428411 B1 EP1428411 B1 EP 1428411B1 EP 01980187 A EP01980187 A EP 01980187A EP 01980187 A EP01980187 A EP 01980187A EP 1428411 B1 EP1428411 B1 EP 1428411B1
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Prior art keywords
audio signal
bandpass
frequency components
filter
bpf
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German (de)
French (fr)
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EP1428411B2 (en
EP1428411A1 (en
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Roland Aubauer
Stefano Ambrosius Klinke
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Gigaset Communications GmbH
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Siemens AG
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    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R3/00—Circuits for transducers, loudspeakers or microphones
    • H04R3/04—Circuits for transducers, loudspeakers or microphones for correcting frequency response

Definitions

  • FIGURE 1 shows a typical frequency response of a small loudspeaker.
  • Electronic audio equipment in which such small electro-acoustic transducers are used and in which consequently the bass reproduction is unsatisfactory, are primarily audio devices (devices for outputting and / or reproducing audio signals) of the communication and information technology as well as consumer electronics and consumer goods, such as Cellular and cordless telephone handsets, notebooks, personal digital assistants, mini-radios, clock radios, portable music players, etc.
  • the perception of a fundamental frequency can be simulated by a combination of harmonics. Therefore, the perception of a low frequency can be simulated with the appropriate combination of its harmonics.
  • the object underlying the invention is to control the bass reproduction of audio signals in electroacoustic transducers based on the psychoacoustic principle referred to as “virtual pitch” or as “residual hearing (hearing of missing fundamental)" such that the perception of the virtual bass reproduction of the audio signals is improved over the prior art.
  • the idea of the invention is to control the reproduction of the low frequencies or basses emitted in the electroacoustic transducer by amplifying the harmonic harmonics already contained in the audio signal in the sense of a simulation that the listener perceives or perceives improved bass reproduction.
  • the control or simulation can be both digitally (claim 1), by a program module in the digital signal processor DSP of the electronic device for output and / or playback of audio signals with the electroacoustic transducer, as well as analog (claim 9), by a Hardware circuit between the digital / analog converter and the power amplifier of the electronic device for output and / or playback of audio signals with the electro-acoustic transducer, done.
  • the program module and the hardware circuitry amplify only the harmonic waves that are above the resonant frequency of the electroacoustic transducer, particularly the loudspeaker, to simulate the perception of the fundamental frequency.
  • Harmonic harmonic extraction is achieved in the program module by bandpass filtering and hardware switching by means of a bandpass filter, while the harmonic gain is controlled by a gain factor in the program module and in the hardware circuitry in a correspondingly designed gain controlled amplifier (English: Gain Controlled Amplifier) expires.
  • the amplification factor is preferably of frequency components of the audio signal is controlled below the resonance frequency or cut-off frequency of the electroacoustic transducer.
  • the advantage of the method according to claim 1 lies in the fact that the amplification of the harmonic original harmonics present in the audio signal ensures a significantly better quality of the modified audio signal generated in the digital signal processor. As a result, in particular distortions of the audio signal are avoided.
  • the inventive method has lower requirements in terms of computer performance and memory requirements in the digital signal processor.
  • the modified audio signal is filtered to amplify selected frequencies.
  • FIG. 2 shows, as a second exemplary embodiment in the form of a functional or block diagram, the speech processing path in a radio FG for outputting and / or reproducing audio signals, in particular speech signals, in which the invention is implemented in a program module PGM of a digital signal processor DSP (digital Implementation).
  • the radio FG receives via an antenna ANT an analog radio signal FS, on which a coded voice information is modulated.
  • a digital demodulated signal DDS is generated from the modulated coded analog radio signal FS.
  • This digital demodulated signal DDS is then supplied to a speech decoder SDK of the digital signal processor DSP.
  • a speech signal or, generally speaking, an audio signal AS is generated from the digital demodulated signal DDS.
  • This audio signal AS is then the program module for controlling the bass reproduction of Audio signals in electroacoustic transducers PGM supplied to the digital signal processor DSP.
  • a modified audio signal MAS is generated from the audio signal AS, which is then further filtered by a filter FIL of the digital signal processor DSP.
  • the filtered modified audio signal MAS is finally applied to a digital-to-analogue converter DAW and then amplified in a power amplifier EVS before the voice information contained in the modified audio signal MAS is output by an electroacoustic transducer EAS, which is preferably designed as a loudspeaker.
  • EAS electroacoustic transducer
  • FIG. 3 shows the speech processing path in the radio FG, in which the invention, unlike FIG. 2, outside the digital signal processor DSP in the analog part of the radio FG in a device for controlling the bass reproduction of audio signals is implemented in electroacoustic transducers STV (analog implementation).
  • the voice signal processing in the radio FG begins with the analogue radio signal FS, on which coded voice information is modulated, being fed via the antenna ANT to the receiver EMP.
  • the digital demodulated signal DDS is again generated by the microprocessor MP and the analog-to-digital converter ADW from the analog radio signal FS.
  • This digital demodulated signal DDS is then fed back to the speech decoder SDK in the digital signal processor DSP.
  • the decoded speech signal or, more generally, the decoded audio signal AS is recovered from the digital demodulated signal DDS.
  • This audio signal AS is then filtered in the filter FIL of the digital signal processor DSP before the filtered audio signal in the digital-to-analog converter DAW is converted accordingly.
  • the converted audio signal AS then becomes the device for controlling the bass reproduction of audio signals in electroacoustic transducers STV supplied, where from the audio signal AS, a modified audio signal MAS is generated.
  • the modified audio signal MAS is subsequently amplified in the power amplifier EVS before the voice information contained in the modified audio signal MAS is output via the electroacoustic converter EAW, which is again preferably designed as a loudspeaker.
  • FIG. 4 shows a first realization form of the program module PGM according to FIG. 2.
  • the audio signal AS is band-pass filtered to isolate a first frequency component FK with a software implemented bandpass filter BPF and low-pass filtered to isolate a second frequency component FK 'with a software implemented low-pass filter TPF. While the first frequency component FK is being amplified, the second frequency component FK 'generates a gain factor VF which determines the gain of the first frequency component FK.
  • the bandpass filter BPF is preferably designed as a finite impulse response (FIR) filter FIR-F or alternatively as an infinite impulse response (IIR) filter IIR-F. If the bandpass filter BPF is a finite impulse response. Filter FIR-F formed, the program module PGM for buffering the audio signal AS a buffer ZWS. This buffer ZWS is not required if the bandpass filter BPF is designed as an Infinite Impulse Response filter IIR-F. To illustrate this in FIGURE 4, the buffer ZWS is shown as a dashed block.
  • the band-pass filtered audio signal FK or the frequency component FK isolated with the bandpass filter BPF is applied to amplify it to the input of a software-implemented amplifier VS controllable by the gain factor VF.
  • a software-implemented amplifier VS controllable by the gain factor VF.
  • software implemented means for calculating signal envelope and / or signal energy MBSE are present in the program module PGM, which supply an input variable for the likewise implemented by software means for calculating the amplification factor MBVF of the program module PGM from the low-pass filtered audio signal FK ' ,
  • the calculation means MBVF then supply the amplification factor VF with which the amplifier VS can be controlled.
  • the modified audio signal MAS is produced, which is preferably filtered to improve the signal quality with a software implemented presence filter PRF.
  • the modified audio signal MAS is supplied to the filter FIL without further filtering by the presence filter PRF.
  • FIG. 5 shows a second embodiment of the program module PGM according to FIG. 2 to isolate the first frequency component FK. While the first frequency component FK is again amplified, the gain of the second frequency component FK 'is again increased first gain FK determining gain factor VF generated.
  • the bandpass filter BPF is again preferably designed as a finite impulse response (FIR) filter FIR-F or alternatively as an infinite impulse response (IIR) filter IIR-F. If the bandpass filter BPF is a finite impulse response. Filter FIR-F formed, the program module PGM again for buffering the audio signal AS the buffer ZWS. This buffer ZWS is not required again when the band-pass filter BPF is designed as an Infinite Impulse Response filter IIR-F. To illustrate this in FIGURE 5, the buffer ZWS is shown as a dashed block.
  • the band-pass filtered audio signal FK or the frequency component FK isolated with the bandpass filter BPF is, as in FIG. 4, applied to its gain to the input of an amplifier VS which can be controlled by the amplification factor VF.
  • the program module PGM again has the means for calculating signal envelope and / or signal energy MBSE, which again supply an input from the low-pass filtered audio signal FK 'to the means for calculating the amplification factor MBVF of the program module PGM.
  • the calculation means MBVF is supplied with a further input variable which originates from further means for calculating signal envelope and / or signal energy MBSE.
  • the further input quantity is calculated by the calculation means MBSE from the unfiltered audio signal AS.
  • the calculation means MBVF then supply from these two input variables the amplification factor VF with which the amplifier VS can be controlled again.
  • the output of the amplifier VS is thus again connected to the band-pass filtered audio signal VSFK amplified by the amplification factor VF.
  • This amplified bandpass filtered audio signal VSFK and the audio signal AS which may have been temporarily stored, are subsequently combined or added again with the aid of the combination means KM of the program module PGM, which are preferably designed again as addition means.
  • the modified audio signal MAS is produced, which is preferably filtered again with the presence filter PRF in order to improve the signal quality.
  • the modified audio signal MAS as explained in the description of FIG. 2, to be supplied to the filter FIL without further filtering by the presence filter PRF.
  • FIG. 6 shows starting from FIG. 4 a third embodiment of the program module PGM according to FIG. 2.
  • the audio signal AS is bandpass filtered again to isolate the first frequency component FK with the bandpass filter BPF and lowpass filtered again to isolate the second frequency component FK 'with the lowpass filter TPF. While the first frequency component FK is again amplified, the amplification factor VF determining the amplification of the first frequency component FK is generated again with the second frequency component FK '.
  • the bandpass filter BPF is again preferably designed as a finite impulse response (FIR) filter FIR-F or alternatively as an infinite impulse response (IIR) filter IIR-F. If the bandpass filter BPF is a finite impulse response. Filter FIR-F formed, the program module PGM again to buffer the audio signal AS the buffer ZWS. This buffer ZWS is not necessary again when the bandpass filter BPF is designed as an Infinite Impulse Response filter IIR-F. To illustrate this in FIGURE 6, the buffer ZWS is shown as a dashed block.
  • the band-pass filtered audio signal FK or the frequency component FK isolated with the bandpass filter BPF is applied to the input of the amplifier VS which can be controlled by the amplification factor VF as shown in FIGS. 4 and 5 for amplification thereof.
  • the program module PGM again has the means for calculating signal envelope and / or signal energy MBSE, which supply an input quantity for means for calculating the amplification factor MBVF of the program module PGM from the low-pass filtered audio signal FK '.
  • the calculation means MBVF is supplied with a further input variable which originates from further means for calculating signal envelope and / or signal energy MBSE.
  • the further input variable is calculated by the calculation means MBSE from the bandpass-filtered audio signal FK.
  • the calculation means MBVF then supply from these two input variables the amplification factor VF with which the amplifier VS can be controlled.
  • the band-pass filtered audio signal VSFK amplified by the amplification factor VF is applied again.
  • This reinforced Bandpass-filtered audio signal VSFK and the audio signal AS which may have been temporarily stored, are subsequently combined or added again with the aid of the combination means KM of the program module PGM, which are preferably designed as addition means.
  • the modified audio signal MAS again arises, which is preferably filtered again to improve the signal quality with the presence filter PRF.
  • the modified audio signal MAS as explained in the description of FIG. 2, to be supplied to the filter FIL without further filtering by the presence filter PRF.
  • FIG. 7 shows an embodiment of the control device STV according to FIG. 3.
  • the audio signal AS is band-pass filtered to isolate the first frequency component FK with a bandpass filter BPF1 designed as a hardware component and to isolate the second frequency component FK 'with a lowpass filter TPF1 designed as a hardware component low-pass filtered. While the first frequency component FK is amplified, the amplification factor VF determining the gain of the first frequency component FK is generated with the second frequency component FK '.
  • the band-pass filtered audio signal FK or the frequency component FK isolated with the bandpass filter BPF1 is applied to amplify it to the input of an amplifier VS1 which can be controlled by the amplification factor VF and is designed as a hardware component.
  • means formed in the control device STV as a hardware module for calculating the signal envelope and / or the signal energy MBSE1 are preferably present consist of the series circuit of a rectifier GLR and another low-pass filter TPF2 and provide from the low-pass filtered audio signal FK 'an input for also designed as a hardware module means for calculating the gain MBVF1 the control device STV.
  • the calculating means MBVF1 then supply the amplification factor VF with which the amplifier VS1 can be controlled.
  • a band-pass filtered audio signal VSFK amplified by the amplification factor VF.
  • This amplified bandpass filtered audio signal VSFK and the audio signal AS are further combined or added with the aid of combination means KM1 of the control device STV, which are preferably designed as an addition means and as a hardware component.
  • the modified audio signal MAS is produced, which is preferably filtered to improve the signal quality with a presence filter PRF1 embodied as a hardware component. But it is also possible that the modified audio signal MAS, as explained in the description of FIGURE 3, without further filtering by the presence filter PRF is supplied to the power amplifier EVS.

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  • Physics & Mathematics (AREA)
  • Engineering & Computer Science (AREA)
  • Acoustics & Sound (AREA)
  • Signal Processing (AREA)
  • Tone Control, Compression And Expansion, Limiting Amplitude (AREA)
  • Circuit For Audible Band Transducer (AREA)

Description

Die Erfindung betrifft ein Verfahren zur Steuerung der Basswiedergabe von Audiosignalen in elektroakustischen Wandlern gemäß dem Oberbegriff des Patentanspruches 1 und eine Vorrichtung zur Steuerung der Basswiedergabe von Audiosignalen in elektroakustischen Wandlern gemäß dem Oberbegriff des Patentanspruches 9.The invention relates to a method for controlling the bass reproduction of audio signals in electroacoustic transducers according to the preamble of claim 1 and an apparatus for controlling the bass reproduction of audio signals in electroacoustic transducers according to the preamble of claim 9.

Die Basswiedergabe von Audiosignalen in einem elektroakustischen Wandler, insbesondere einem Lautsprecher oder einer Hörkapsel, ist durch die Größe des elektroakustischen Wandlers, des Lautsprechers bzw. der Hörkapsel bedingt. Je kleiner die Lautsprecher-Membrane und deren maximale Auslenkung sind, desto höher ist die untere Resonanzfrequenz.The bass reproduction of audio signals in an electroacoustic transducer, in particular a loudspeaker or an earphone, is determined by the size of the electroacoustic transducer, the loudspeaker or the earphone. The smaller the speaker diaphragm and its maximum deflection, the higher the lower resonant frequency.

In FIGUR 1 ist ein typischer Frequenzgang eines kleinen Lautsprechers dargestellt. Elektronische Audiogeräte, in denen solche kleinen elektroakustischen Wandler zum Einsatz kommen und in denen folglich die Basswiedergabe unbefriedigend ist, sind in erster Linie Audiogeräte (Geräte zur Aus- und/oder Wiedergabe von Audiosignalen) der Kommunikations- und Informationstechnik sowie der Unterhaltungs- und Konsumgüterelektronik, wie z.B. Mobilfunk- und Schnurlostelefon-Handapparate, Notebooks, Personal Digital Assistants, Mini-Radios, Radiowecker, tragbare Musikabspielgeräte etc.FIGURE 1 shows a typical frequency response of a small loudspeaker. Electronic audio equipment in which such small electro-acoustic transducers are used and in which consequently the bass reproduction is unsatisfactory, are primarily audio devices (devices for outputting and / or reproducing audio signals) of the communication and information technology as well as consumer electronics and consumer goods, such as Cellular and cordless telephone handsets, notebooks, personal digital assistants, mini-radios, clock radios, portable music players, etc.

Um die Basswiedergabe mit einem kleinen Lautsprecher zu verbessern, kann ein bekannter psychoakustisches Prinzip benutzt werden. Dieses Prinzip wird als "Residual Hearing (Hearing of Missing Fundamentals)" oder als "Virtual Pitch" bezeichnet.To improve the bass reproduction with a small speaker, a well-known psychoacoustic principle can be used. This principle is referred to as "residual hearing (Hearing of Missing Fundamentals)" or "virtual pitch".

Nach diesem Prinzip kann die Wahrnehmung einer Grundfrequenz durch eine Kombination von Oberwellen simuliert werden. Daher kann auch die Wahrnehmung einer tiefen Frequenz mit der entsprechenden Kombination ihrer Oberwellen simuliert werden.According to this principle, the perception of a fundamental frequency can be simulated by a combination of harmonics. Therefore, the perception of a low frequency can be simulated with the appropriate combination of its harmonics.

Eine detaillierte Beschreibung des Virtual Pitch"-Prinzips ist in der Publikation " Psychoakustik" von E. Zwicker; H.Fastl; Springer Verlag , 2nd. Edition, 1999 zu finden.A detailed description of the Virtual Pitch Principle is in the publication " Psychoacoustics "by E. Zwicker, H.Fastl, Springer Verlag, 2nd Edition, 1999 to find.

Aus der US 6,111,960 und der US 5,930,373 sind auf dem psychoakustischen Prinzip beruhende Verfahren bekannt, die anhand des Audiosignals eine entsprechende Reihe von Oberwellen erzeugen, um die Frequenzen unterhalb der Grenzfrequenz zu simulieren.From the US 6,111,960 and the US 5,930,373 For example, methods based on the psychoacoustic principle are known which generate a corresponding series of harmonics from the audio signal to simulate the frequencies below the cutoff frequency.

Aus der WO 00/15003 ist ein auf dem psychoakustischen Prinzip beruhendes Verfahren bekannt, bei dem die in dem Audiosignal vorhandenen Oberwellen verstärkt werden. Dabei werden zur Verbesserung der Basswiedergabe der Audiosignale in elektroakustischen Wandlern tiefe Frequenzkomponenten des Audiosignals zu einem tieffrequenten Audiosignal isoliert, die isolierten tiefen Frequenzkomponenten mit einer Vielzahl von Bandpassfiltern gefiltert, die bandpassgefilterten Frequenzkomponenten in einem bezüglich des Verstärkungsfaktors steuerbaren Verstärker verstärkt, wobei der Verstärkungsfaktor aus der Einhüllenden der bandpassgefilterten Frequenzkomponenten gewonnen wird, und ein simuliertes tieffrequentes Audiosignal durch Kombinieren des ursprünglichen Audiosignals mit den verstärkten Frequenzkomponenten erzeugt.From the WO 00/15003 A method based on the psychoacoustic principle is known in which the harmonics present in the audio signal are amplified. In order to improve the bass reproduction of the audio signals in electroacoustic transducers, low frequency components of the audio signal are isolated into a low-frequency audio signal, the isolated low frequency components filtered with a plurality of bandpass filters, the bandpass filtered frequency components amplified in a gain-controllable amplifier, the amplification factor from the envelope of the band-pass filtered frequency components, and generates a simulated low-frequency audio signal by combining the original audio signal with the amplified frequency components.

Die der Erfindung zugrundeliegende Aufgabe besteht darin, die Basswiedergabe von Audiosignalen in elektroakustischen Wandlern basierend auf dem als "virtual pitch" oder als "residual hearing (hearing of missing fundamental)" bezeichneten psychoakustischen Prinzip so zu steuern, dass die Wahrnehmung der virtuellen Basswiedergabe der Audiosignale gegenüber dem Stand der Technik verbessert ist.The object underlying the invention is to control the bass reproduction of audio signals in electroacoustic transducers based on the psychoacoustic principle referred to as "virtual pitch" or as "residual hearing (hearing of missing fundamental)" such that the perception of the virtual bass reproduction of the audio signals is improved over the prior art.

Diese Aufgabe wird sowohl ausgehend von dem im Oberbegriff des Patentanspruches 1 definierten Verfahren durch die im Kennzeichen des Patentanspruches 1 angegebenen Merkmale als auch ausgehend von der im Oberbegriff des Patentanspruches 9 definierten Vorrichtung durch die im Kennzeichen des Patentanspruches 9 angegebenen Merkmale gelöst.This object is achieved both on the basis of the method defined in the preamble of claim 1 by the features specified in the characterizing part of claim 1 as well as starting from the device defined in the preamble of claim 9 by the features specified in the characterizing part of claim 9.

Die die Erfindung ausmachende Idee besteht darin, die Wiedergabe der in dem elektroakustischen Wandler abgegebenen tiefen Frequenzen bzw. Bässe durch das Verstärken der schon im Audiosignal enthaltenen harmonischen Oberwellen so im Sinne einer Simulation zu steuern, dass der Hörer eine verbesserte Basswiedergabe empfindet bzw. wahrnimmt. Die Steuerung bzw. Simulation kann dabei sowohl digital (Anspruch 1), durch ein Programmmodul im Digitalen Signal-Prozessor DSP des elektronischen Gerätes zur Aus- und/oder Wiedergabe von Audiosignalen mit dem elektroakustischen Wandler, als auch analog (Anspruch 9), durch eine Hardware-Schaltung zwischen dem Digital/Analog-Wandler und dem Endverstärker des elektronischen Gerätes zur Aus- und/oder Wiedergabe von Audiosignalen mit dem elektroakustischen Wandler, erfolgen.The idea of the invention is to control the reproduction of the low frequencies or basses emitted in the electroacoustic transducer by amplifying the harmonic harmonics already contained in the audio signal in the sense of a simulation that the listener perceives or perceives improved bass reproduction. The control or simulation can be both digitally (claim 1), by a program module in the digital signal processor DSP of the electronic device for output and / or playback of audio signals with the electroacoustic transducer, as well as analog (claim 9), by a Hardware circuit between the digital / analog converter and the power amplifier of the electronic device for output and / or playback of audio signals with the electro-acoustic transducer, done.

Mit dem Programmmodul und der Hardware-Schaltung werden nur die harmonischen Oberwellen verstärkt, die sich oberhalb der Resonanzfrequenz des elektroakustischen Wandlers, insbesondere des Lautsprechers, befinden, um die Wahrnehmung der Grundfrequenz zu simulieren. Die Extraktion bzw. Isolierung der harmonischen Oberwellen wird beim Programmmodul durch Bandpassfilterung und bei der Hardware-Schaltung mittels eines Bandpassfilters erreicht, während die Verstärkung der Oberwellen gesteuert durch einen Verstärkungsfaktor in dem Programmmodul softwaregestützt und in der Hardware-Schaltung in einem dafür entsprechend ausgebildeten verstärkungsfaktorgesteuerten Verstärker (engl.: Gain Controlled Amplifier) abläuft. Der Verstärkungsfaktor wird vorzugsweise von Frequenzkomponenten des Audiosignals unterhalb der Resonanzfrequenz bzw. Grenzfrequenz des elektroakustischen Wandlers gesteuert.The program module and the hardware circuitry amplify only the harmonic waves that are above the resonant frequency of the electroacoustic transducer, particularly the loudspeaker, to simulate the perception of the fundamental frequency. Harmonic harmonic extraction is achieved in the program module by bandpass filtering and hardware switching by means of a bandpass filter, while the harmonic gain is controlled by a gain factor in the program module and in the hardware circuitry in a correspondingly designed gain controlled amplifier (English: Gain Controlled Amplifier) expires. The amplification factor is preferably of frequency components of the audio signal is controlled below the resonance frequency or cut-off frequency of the electroacoustic transducer.

Der Vorteil des Verfahrens gemäß Anspruch 1 liegt darin, dass die Verstärkung der im Audiosignal vorhandenen harmonischen Original-Oberwellen eine deutliche bessere Qualität des im Digitalen Signal-Prozessor erzeugten modifizierten Audiosignals gewährleistet. Dadurch werden insbesondere Verzerrungen des Audiosignals vermieden. Außerdem stellt das erfindungsgemäße Verfahren geringere Anforderungen hinsichtlich der Rechnerleistung und des Speicherbedarfs im Digitalen Signal-Prozessor.The advantage of the method according to claim 1 lies in the fact that the amplification of the harmonic original harmonics present in the audio signal ensures a significantly better quality of the modified audio signal generated in the digital signal processor. As a result, in particular distortions of the audio signal are avoided. In addition, the inventive method has lower requirements in terms of computer performance and memory requirements in the digital signal processor.

Vorteilhafte Weiterbildungen der Erfindung sind in den Unteransprüchen angegeben.Advantageous developments of the invention are specified in the subclaims.

So ist es nach Anspruch 2 iVm Anspruch 4 von Vorteil, wenn bei der Verwendung eines "Finite Impulse Response"-Filters - im Unterschied zu der Verwendung eines "Infinite Impulse Response"-Filter gemäß Anspruch 3 - das mit den verstärkten Frequenzkomponenten zu kombinierende Audiosignal gepuffert wird, um für die Kombination aufgrund der Verwendung des FIR-Filters vorhandene Phasenverschiebungen zwischen der verstärkten Frequenzkomponenten und dem Audiosignal zu kompensieren.Thus, it is advantageous according to claim 2 in conjunction with claim 4, when using a "Finite Impulse Response" filter - in contrast to the use of an "Infinite Impulse Response" filter according to claim 3 - the audio signal to be combined with the amplified frequency components is buffered to compensate for the combination due to the use of the FIR filter existing phase shifts between the amplified frequency components and the audio signal.

Nach den Ansprüchen 7 und 10 ist es vorteilhaft, wenn zur Verbesserung der Qualität des vom elektroakustischen Wandler abgegebenen modifizierten Audiosignals das modifizierte Audiosignal zur Verstärkung von ausgewählten Frequenzen gefiltert wird.According to claims 7 and 10, it is advantageous if, in order to improve the quality of the modified audio signal output by the electroacoustic transducer, the modified audio signal is filtered to amplify selected frequencies.

Zwei Ausführungsbeispiele der Erfindung werden anhand der FIGUREN 2 bis 7 erläutert. Es zeigen:

  • FIGUR 2 die digitale Implementierung des erfindungsgemäßen Verfahrens in Form eines Programmmoduls in einem Digitalen Signal-Prozessor eines elektronischen Funkgerätes zur Aus- und/oder Wiedergabe von Audiosignalen,
  • FIGUR 3 die analoge Implementierung der erfindungsgemäßen Vorrichtung in das Hardware-Konzept eines elektronischen Funkgerätes zur Aus- und/oder Wiedergabe von Audiosignalen,
  • FIGUR 4 eine erste Realisierungsform des Programmmoduls nach FIGUR 2,
  • FIGUR 5 eine zweite Realisierungsform des Programmmoduls nach FIGUR 2,
  • FIGUR 6 eine dritte Realisierungsform des Programmmoduls nach FIGUR 2,
  • FIGUR 7 eine Realisierungsform der Steuerungsvorrichtung nach FIGUR 3.
Two embodiments of the invention will be explained with reference to FIGS 2 to 7. Show it:
  • FIGURE 2 shows the digital implementation of the inventive method in the form of a program module in a digital Signal processor of an electronic radio device for outputting and / or reproducing audio signals,
  • FIG. 3 shows the analog implementation of the device according to the invention in the hardware concept of an electronic radio device for outputting and / or reproducing audio signals.
  • FIG. 4 shows a first realization form of the program module according to FIG. 2,
  • FIG. 5 shows a second embodiment of the program module according to FIG. 2,
  • FIG. 6 shows a third embodiment of the program module according to FIG. 2,
  • FIG. 7 shows an embodiment of the control device according to FIG. 3.

FIGUR 2 zeigt als zweites Ausführungsbeispiel in Form eines Funktions- oder Blockschaltbildes die Sprachverarbeitungsstrecke in einem Funkgerät FG zur Aus- und/oder Wiedergabe von Audiosignalen, insbesondere Sprachsignalen, bei dem die Erfindung in einem Programmmodul PGM eines Digitalen Signal-Prozessors DSP implementiert ist (digitale Implementierung). Das Funkgerät FG empfängt über eine Antenne ANT ein analoges Funksignal FS, auf dem eine kodierte Sprachinformation aufmoduliert ist. In einem Empfänger EMP unterstützt von einem Mikroprozessor MP und einem Analog-Digital-Wandler ADW wird aus dem modulierten codierten analogen Funksignal FS ein digitales demoduliertes Signal DDS erzeugt. Dieses digitale demodulierte Signal DDS wird danach einem Sprachdekodierer SDK des Digitalen Signal-Prozessors DSP zugeführt. In dem Sprachdekodierer SDK wird aus dem digitalen demodulierten Signal DDS ein Sprachsignal oder - ganz allgemein formuliert - ein Audiosignal AS erzeugt. Dieses Audiosignal AS wird anschließend dem Programmmodul zur Steuerung der Basswiedergabe von Audiosignalen in elektroakustischen Wandlern PGM des Digitalen Signal-Prozessors DSP zugeführt. In dem Programmmodul PGM des digitalen Signal-Prozessors DSP wird aus dem Audiosignal AS ein modifiziertes Audiosignal MAS generiert, das dann im weiteren von einem Filter FIL des Digitalen Signal-Prozessors DSP gefiltert wird. Das gefilterte modifizierte Audiosignal MAS wird schließlich auf einen Digital-Analog-Wandler DAW gegeben und danach in einem Endverstärker EVS verstärkt, bevor die in dem modifizierten Audiosignal MAS enthaltene Sprachinformation von einem elektroakustischen Wandler EAS, der vorzugsweise als Lautsprecher ausgebildet ist, ausgegeben wird.FIG. 2 shows, as a second exemplary embodiment in the form of a functional or block diagram, the speech processing path in a radio FG for outputting and / or reproducing audio signals, in particular speech signals, in which the invention is implemented in a program module PGM of a digital signal processor DSP (digital Implementation). The radio FG receives via an antenna ANT an analog radio signal FS, on which a coded voice information is modulated. In a receiver EMP supported by a microprocessor MP and an analog-to-digital converter ADW, a digital demodulated signal DDS is generated from the modulated coded analog radio signal FS. This digital demodulated signal DDS is then supplied to a speech decoder SDK of the digital signal processor DSP. In the speech decoder SDK, a speech signal or, generally speaking, an audio signal AS is generated from the digital demodulated signal DDS. This audio signal AS is then the program module for controlling the bass reproduction of Audio signals in electroacoustic transducers PGM supplied to the digital signal processor DSP. In the program module PGM of the digital signal processor DSP, a modified audio signal MAS is generated from the audio signal AS, which is then further filtered by a filter FIL of the digital signal processor DSP. The filtered modified audio signal MAS is finally applied to a digital-to-analogue converter DAW and then amplified in a power amplifier EVS before the voice information contained in the modified audio signal MAS is output by an electroacoustic transducer EAS, which is preferably designed as a loudspeaker.

FIGUR 3 zeigt als zweites Ausführungsbeispiel in Form eines Funktions- oder Blockschaltbildes die Sprachverarbeitungsstrecke in dem Funkgerät FG, bei dem die Erfindung im Unterschied zu FIGUR 2 außerhalb des Digitalen Signal-Prozessors DSP im Analogteil des Funkgerätes FG in einer Vorrichtung zur Steuerung des Basswiedergabe von Audiosignalen in elektroakustischen Wandlern STV implementiert ist (analoge Implementierung). Die Sprachsignalverarbeitung in dem Funkgerät FG beginnt wiederum damit, dass das analoge Funksignal FS, auf dem eine kodierte Sprachinformation aufmoduliert ist, über die Antenne ANT dem Empfänger EMP zugeführt wird. In dem Empfänger EMP wird wiederum unterstützt durch den Mikroprozessor MP und den Analog-Digital-Wandler ADW aus dem analogen Funksignal FS wiederum das digitale demodulierte Signal DDS erzeugt. Dieses digitale demodulierte Signal DDS wird anschließend wieder dem Sprachdekodierer SDK in dem Digitalen Signal-Prozessor DSP zugeführt. In dem Sprachdecodierer SDK wird aus dem digitalen demodulierten Signal DDS wieder das dekodierte Sprachsignal oder ganz allgemein das dekodierte Audiosignal AS gewonnen. Dieses Audiosignal AS wird anschließend in dem Filter FIL des Digitalen Signal-Prozessors DSP gefiltert, bevor das gefilterte Audiosignal in dem Digital-Analog-Wandler DAW entsprechend gewandelt wird. Das gewandelte Audiosignal AS wird anschließend der Vorrichtung zur Steuerung der Basswiedergabe von Audiosignalen in elektroakustischen Wandlern STV zugeführt, wo aus dem Audiosignal AS ein modifiziertes Audiosignal MAS generiert wird. Das modifizierte Audiosignal MAS wird im Anschluss daran in dem Endverstärker EVS verstärkt, bevor die in dem modifizierten Audiosignal MAS enthaltene Sprachinformation über den elektroakustischen Wandler EAW, der wieder vorzugsweise als Lautsprecher ausgebildet ist, ausgegeben wird.As a second exemplary embodiment in the form of a functional or block diagram, FIG. 3 shows the speech processing path in the radio FG, in which the invention, unlike FIG. 2, outside the digital signal processor DSP in the analog part of the radio FG in a device for controlling the bass reproduction of audio signals is implemented in electroacoustic transducers STV (analog implementation). The voice signal processing in the radio FG, in turn, begins with the analogue radio signal FS, on which coded voice information is modulated, being fed via the antenna ANT to the receiver EMP. In turn, in the receiver EMP, the digital demodulated signal DDS is again generated by the microprocessor MP and the analog-to-digital converter ADW from the analog radio signal FS. This digital demodulated signal DDS is then fed back to the speech decoder SDK in the digital signal processor DSP. In the speech decoder SDK, the decoded speech signal or, more generally, the decoded audio signal AS is recovered from the digital demodulated signal DDS. This audio signal AS is then filtered in the filter FIL of the digital signal processor DSP before the filtered audio signal in the digital-to-analog converter DAW is converted accordingly. The converted audio signal AS then becomes the device for controlling the bass reproduction of audio signals in electroacoustic transducers STV supplied, where from the audio signal AS, a modified audio signal MAS is generated. The modified audio signal MAS is subsequently amplified in the power amplifier EVS before the voice information contained in the modified audio signal MAS is output via the electroacoustic converter EAW, which is again preferably designed as a loudspeaker.

FIGUR 4 zeigt eine erste Realisierungsform des Programmmoduls PGM gemäß der FIGUR 2. Das Audiosignal AS wird zur Isolation einer ersten Frequenzkomponente FK mit einem mittels Software realisierten Bandpassfilter BPF bandpassgefiltert und zur Isolation einer zweiten Frequenzkomponente FK' mit einem mittels Software realisierten Tiefpassfilter TPF tiefpassgefiltert. Während die erste Frequenzkomponente FK verstärkt wird, wird mit der zweiten Frequenzkomponente FK' ein die Verstärkung der ersten Frequenzkomponente FK bestimmender Verstärkungsfaktor VF erzeugt.FIG. 4 shows a first realization form of the program module PGM according to FIG. 2. The audio signal AS is band-pass filtered to isolate a first frequency component FK with a software implemented bandpass filter BPF and low-pass filtered to isolate a second frequency component FK 'with a software implemented low-pass filter TPF. While the first frequency component FK is being amplified, the second frequency component FK 'generates a gain factor VF which determines the gain of the first frequency component FK.

Anstelle des Tiefpassfilters TPF kann alternativ auch ein weiteres mittels Software realisiertes Bandpassfilter oder sogar das die erste Frequenzkomponente FK erzeugende Bandpassfilter BPF verwendet werden. Im letztgenannten Fall wären die beiden Frequenzkomponenten FK, FK' gleich (FK=FK').Instead of the low-pass filter TPF, another band-pass filter realized by software or even the bandpass filter BPF generating the first frequency component FK can alternatively also be used. In the latter case, the two frequency components FK, FK 'would be the same (FK = FK').

Das Bandpassfilter BPF ist vorzugsweise als Finite Impulse Response"-Filter (FIR-Filter) FIR-F oder alternativ als "Infinite Impulse Response "-Filter (IIR-Filter) IIR-F ausgebildet. Ist das Bandpassfilter BPF als Finite Impulse Response"-Filter FIR-F ausgebildet, enthält das Programmmodul PGM zur Pufferung des Audiosignals AS einen Zwischenspeicher ZWS. Dieser Zwischenspeicher ZWS ist dann, wenn das Bandpassfilter BPF als Infinite Impulse Response"-Filter IIR-F ausgebildet ist, nicht erforderlich. Um dieses zu in der FIGUR 4 darzustellen, ist der Zwischenspeicher ZWS als gestrichelter Block dargestellt.The bandpass filter BPF is preferably designed as a finite impulse response (FIR) filter FIR-F or alternatively as an infinite impulse response (IIR) filter IIR-F. If the bandpass filter BPF is a finite impulse response. Filter FIR-F formed, the program module PGM for buffering the audio signal AS a buffer ZWS. This buffer ZWS is not required if the bandpass filter BPF is designed as an Infinite Impulse Response filter IIR-F. To illustrate this in FIGURE 4, the buffer ZWS is shown as a dashed block.

Das bandpassgefilterte Audiosignal FK bzw. die mit dem Bandpassfilter BPF isolierte Frequenzkomponente FK wird zur deren Verstärkung an den Eingang eines mit dem Verstärkungsfaktor VF steuerbaren mittels Software realisierten Verstärker VS gelegt. Für die Ermittlung des Verstärkungsfaktor VF sind in dem Programmmodul PGM mittels Software realisierte Mittel zur Berechnung von Signaleinhüllende und/oder Signalenergie MBSE vorhanden, die aus dem tiefpassgefilterten Audiosignal FK' eine Eingangsgröße für ebenfalls mittels Software realisierte Mittel zur Berechnung des Verstärkungsfaktors MBVF des Programmmoduls PGM liefern. Die Berechnungsmittel MBVF liefern dann den Verstärkungsfaktor VF, mit dem der Verstärker VS steuerbar ist. Am Ausgang des Verstärkers VS liegt somit ein mit dem Verstärkungsfaktor VF verstärktes bandpassgefiltertes Audiosignal VSFK an. Dieses verstärkte bandpassgefilterte Audiosignal VSFK und das Audiosignal AS, das gegebenenfalls zwischengespeichert worden ist, werden im weiteren mit Hilfe von vorzugsweise als Additionsmittel ausgebildeten, mittels Software realisierten Kombinationsmittel KM des Programmmoduls PGM kombiniert bzw. addiert. Infolge dieser Operation entsteht das modifizierte Audiosignal MAS, das vorzugsweise zur Verbesserung der Signalqualität mit einem mittels Software realisierten Präsenzfilter PRF gefiltert wird. Es ist aber auch möglich, dass das modifizierte Audiosignal MAS, wie bei der Beschreibung der FIGUR 2 erläutert, ohne weitere Filterung durch das Präsenzfilter PRF dem Filter FIL zugeführt wird.The band-pass filtered audio signal FK or the frequency component FK isolated with the bandpass filter BPF is applied to amplify it to the input of a software-implemented amplifier VS controllable by the gain factor VF. For the determination of the amplification factor VF, software implemented means for calculating signal envelope and / or signal energy MBSE are present in the program module PGM, which supply an input variable for the likewise implemented by software means for calculating the amplification factor MBVF of the program module PGM from the low-pass filtered audio signal FK ' , The calculation means MBVF then supply the amplification factor VF with which the amplifier VS can be controlled. At the output of the amplifier VS there is thus a band-pass filtered audio signal VSFK amplified by the amplification factor VF. This amplified bandpass filtered audio signal VSFK and the audio signal AS, which may have been buffered, are further combined or added with the aid of combination means KM of the program module PGM, which are preferably designed as addition means and implemented by means of software. As a result of this operation, the modified audio signal MAS is produced, which is preferably filtered to improve the signal quality with a software implemented presence filter PRF. However, it is also possible that the modified audio signal MAS, as explained in the description of FIG. 2, is supplied to the filter FIL without further filtering by the presence filter PRF.

FIGUR 5 zeigt ausgehend von FIGUR 4 eine zweite Realisierungsform des Programmmoduls PGM gemäß der FIGUR 2. Das Audiosignal AS wird zur Isolation der ersten Frequenzkomponente FK wieder mit dem Bandpassfilter BPF bandpassgefiltert und zur Isolation der zweiten Frequenzkomponente FK' mit dem Tiefpassfilter TPF tiefpassgefiltert. Während die erste Frequenzkomponente FK wieder verstärkt wird, wird mit der zweiten Frequenzkomponente FK' wieder der die Verstärkung der ersten Frequenzkomponente FK bestimmende Verstärkungsfaktor VF erzeugt.FIG. 5 shows a second embodiment of the program module PGM according to FIG. 2 to isolate the first frequency component FK. While the first frequency component FK is again amplified, the gain of the second frequency component FK 'is again increased first gain FK determining gain factor VF generated.

Anstelle des Tiefpassfilters TPF kann wiederum alternativ auch ein weiteres Bandpassfilter oder sogar das die erste Frequenzkomponente FK erzeugende Bandpassfilter BPF verwendet werden. Im letztgenannten Fall wären die beiden Frequenzkomponenten FK, FK' dann wieder gleich (FK=FK').Instead of the low-pass filter TPF, alternatively, another band-pass filter or even the bandpass filter BPF generating the first frequency component FK may alternatively be used. In the latter case, the two frequency components FK, FK 'would then be equal again (FK = FK').

Das Bandpassfilter BPF ist wieder vorzugsweise als Finite Impulse Response "-Filter (FIR-Filter) FIR-F oder alternativ als "Infinite Impulse Response "-Filter (IIR-Filter) IIR-F ausgebildet. Ist das Bandpassfilter BPF als Finite Impulse Response "-Filter FIR-F ausgebildet, enthält das Programmmodul PGM wieder zur Pufferung des Audiosignals AS den Zwischenspeicher ZWS. Dieser Zwischenspeicher ZWS ist dann wieder, wenn das Bandpassfilter BPF als Infinite Impulse Response"-Filter IIR-F ausgebildet ist, nicht erforderlich. Um dieses zu in der FIGUR 5 darzustellen, ist der Zwischenspeicher ZWS als gestrichelter Block dargestellt.The bandpass filter BPF is again preferably designed as a finite impulse response (FIR) filter FIR-F or alternatively as an infinite impulse response (IIR) filter IIR-F. If the bandpass filter BPF is a finite impulse response. Filter FIR-F formed, the program module PGM again for buffering the audio signal AS the buffer ZWS. This buffer ZWS is not required again when the band-pass filter BPF is designed as an Infinite Impulse Response filter IIR-F. To illustrate this in FIGURE 5, the buffer ZWS is shown as a dashed block.

Das bandpassgefilterte Audiosignal FK bzw. die mit dem Bandpassfilter BPF isolierte Frequenzkomponente FK wird wie in der FIGUR 4 zur deren Verstärkung an den Eingang eines mit dem Verstärkungsfaktor VF steuerbaren Verstärker VS gelegt. Für die Ermittlung des Verstärkungsfaktor VF sind in dem Programmmodul PGM wieder die Mittel zur Berechnung von Signaleinhüllende und/oder Signalenergie MBSE vorhanden, die aus dem tiefpassgefilterten Audiosignal FK' wieder eine Eingangsgröße für die Mittel zur Berechnung des Verstärkungsfaktors MBVF des Programmmoduls PGM liefern.The band-pass filtered audio signal FK or the frequency component FK isolated with the bandpass filter BPF is, as in FIG. 4, applied to its gain to the input of an amplifier VS which can be controlled by the amplification factor VF. For the determination of the amplification factor VF, the program module PGM again has the means for calculating signal envelope and / or signal energy MBSE, which again supply an input from the low-pass filtered audio signal FK 'to the means for calculating the amplification factor MBVF of the program module PGM.

In der Realisierungsform des Programmmoduls PGM gemäß der FIGUR 5 wird im Unterschied zu der gemäß der FIGUR 4 den Berechnungsmitteln MBVF eine weitere Eingangsgröße zugeführt, die von weiteren Mitteln zur Berechnung von Signaleinhüllende und/oder Signalenergie MBSE stammt. Die weitere Eingangsgröße wird von den Berechnungsmitteln MBSE aus dem ungefilterten Audiosignal AS berechnet.In the realization form of the program module PGM according to FIG. 5, in contrast to that according to FIG. 4, the calculation means MBVF is supplied with a further input variable which originates from further means for calculating signal envelope and / or signal energy MBSE. The further input quantity is calculated by the calculation means MBSE from the unfiltered audio signal AS.

Die Berechnungsmittel MBVF liefern dann aus diesen beiden Eingangsgrößen den Verstärkungsfaktor VF, mit dem der Verstärker VS wieder steuerbar ist. Am Ausgang des Verstärkers VS liegt somit wieder das mit dem Verstärkungsfaktor VF verstärkte bandpassgefilterte Audiosignal VSFK an. Dieses verstärkte bandpassgefilterte Audiosignal VSFK und das Audiosignal AS, das gegebenenfalls zwischengespeichert worden ist, werden im weiteren wieder mit Hilfe der vorzugsweise wieder als Additionsmittel ausgebildeten Kombinationsmittel KM des Programmmoduls PGM kombiniert bzw. addiert. Infolge dieser Operation entsteht das modifizierte Audiosignal MAS, das vorzugsweise zur Verbesserung der Signalqualität wieder mit dem Präsenzfilter PRF gefiltert wird. Es ist aber auch wieder möglich, dass das modifizierte Audiosignal MAS, wie bei der Beschreibung der FIGUR 2 erläutert, ohne weitere Filterung durch das Präsenzfilter PRF dem Filter FIL zugeführt wird.The calculation means MBVF then supply from these two input variables the amplification factor VF with which the amplifier VS can be controlled again. The output of the amplifier VS is thus again connected to the band-pass filtered audio signal VSFK amplified by the amplification factor VF. This amplified bandpass filtered audio signal VSFK and the audio signal AS, which may have been temporarily stored, are subsequently combined or added again with the aid of the combination means KM of the program module PGM, which are preferably designed again as addition means. As a result of this operation, the modified audio signal MAS is produced, which is preferably filtered again with the presence filter PRF in order to improve the signal quality. However, it is also possible again for the modified audio signal MAS, as explained in the description of FIG. 2, to be supplied to the filter FIL without further filtering by the presence filter PRF.

FIGUR 6 zeigt ausgehend von FIGUR 4 eine dritte Realisierungsform des Programmmoduls PGM gemäß der FIGUR 2. Das Audiosignal AS wird zur Isolation der ersten Frequenzkomponente FK erneut mit dem Bandpassfilter BPF bandpassgefiltert und zur Isolation der zweiten Frequenzkomponente FK' erneut mit dem Tiefpassfilter TPF tiefpassgefiltert. Während die erste Frequenzkomponente FK wieder verstärkt wird, wird mit der zweiten Frequenzkomponente FK' erneut der die Verstärkung der ersten Frequenzkomponente FK bestimmende Verstärkungsfaktor VF erzeugt.FIG. 6 shows starting from FIG. 4 a third embodiment of the program module PGM according to FIG. 2. The audio signal AS is bandpass filtered again to isolate the first frequency component FK with the bandpass filter BPF and lowpass filtered again to isolate the second frequency component FK 'with the lowpass filter TPF. While the first frequency component FK is again amplified, the amplification factor VF determining the amplification of the first frequency component FK is generated again with the second frequency component FK '.

Anstelle des Tiefpassfilters TPF kann erneut alternativ auch ein weiteres Bandpassfilter oder sogar das die erste Frequenzkomponente FK erzeugende Bandpassfilter BPF verwendet werden. Im letztgenannten Fall wären die beiden Frequenzkomponenten FK, FK' gleich (FK=FK').Instead of the low-pass filter TPF, alternatively, another band-pass filter or even the bandpass filter BPF generating the first frequency component FK may also be used. In the latter case, the two frequency components FK, FK 'would be the same (FK = FK').

Das Bandpassfilter BPF ist erneut vorzugsweise als Finite Impulse Response"-Filter (FIR-Filter) FIR-F oder alternativ als "Infinite Impulse Response"-Filter (IIR-Filter) IIR-F ausgebildet. Ist das Bandpassfilter BPF als Finite Impulse Response"-Filter FIR-F ausgebildet, enthält das Programmmodul PGM erneut zur Pufferung des Audiosignals AS den Zwischenspeicher ZWS. Dieser Zwischenspeicher ZWS ist dann erneut, wenn das Bandpassfilter BPF als Infinite Impulse Response "-Filter IIR-F ausgebildet ist, nicht erforderlich. Um dieses zu in der FIGUR 6 darzustellen, ist der Zwischenspeicher ZWS als gestrichelter Block dargestellt.The bandpass filter BPF is again preferably designed as a finite impulse response (FIR) filter FIR-F or alternatively as an infinite impulse response (IIR) filter IIR-F. If the bandpass filter BPF is a finite impulse response. Filter FIR-F formed, the program module PGM again to buffer the audio signal AS the buffer ZWS. This buffer ZWS is not necessary again when the bandpass filter BPF is designed as an Infinite Impulse Response filter IIR-F. To illustrate this in FIGURE 6, the buffer ZWS is shown as a dashed block.

Das bandpassgefilterte Audiosignal FK bzw. die mit dem Bandpassfilter BPF isolierte Frequenzkomponente FK wird wie in den FIGUREN 4 und 5 zur deren Verstärkung an den Eingang des mit dem Verstärkungsfaktor VF steuerbaren Verstärker VS gelegt. Für die Ermittlung des Verstärkungsfaktor VF sind in dem Programmmodul PGM erneut die Mittel zur Berechnung von Signaleinhüllende und/oder Signalenergie MBSE vorhanden, die aus dem tiefpassgefilterten Audiosignal FK' eine Eingangsgröße für Mittel zur Berechnung des Verstärkungsfaktors MBVF des Programmmoduls PGM liefern.The band-pass filtered audio signal FK or the frequency component FK isolated with the bandpass filter BPF is applied to the input of the amplifier VS which can be controlled by the amplification factor VF as shown in FIGS. 4 and 5 for amplification thereof. For the determination of the amplification factor VF, the program module PGM again has the means for calculating signal envelope and / or signal energy MBSE, which supply an input quantity for means for calculating the amplification factor MBVF of the program module PGM from the low-pass filtered audio signal FK '.

In der Realisierungsform des Programmmoduls PGM gemäß der FIGUR 6 wird im Unterschied zu der gemäß der FIGUR 4 den Berechnungsmitteln MBVF eine weitere Eingangsgröße zugeführt, die von weiteren Mitteln zur Berechnung von Signaleinhüllende und/oder Signalenergie MBSE stammt. Die weitere Eingangsgröße wird im Unterschied zu der gemäß der FIGUR 5 von den Berechnungsmitteln MBSE aus dem bandpassgefilterten Audiosignal FK berechnet.In the realization form of the program module PGM according to FIG. 6, in contrast to that according to FIG. 4, the calculation means MBVF is supplied with a further input variable which originates from further means for calculating signal envelope and / or signal energy MBSE. In contrast to that according to FIG. 5, the further input variable is calculated by the calculation means MBSE from the bandpass-filtered audio signal FK.

Die Berechnungsmittel MBVF liefern dann aus diesen beiden Eingangsgrößen den Verstärkungsfaktor VF, mit dem der Verstärker VS steuerbar ist. Am Ausgang des Verstärkers VS liegt somit erneut das mit dem Verstärkungsfaktor VF verstärkte bandpassgefilterte Audiosignal VSFK an. Dieses verstärkte bandpassgefilterte Audiosignal VSFK und das Audiosignal AS, das gegebenenfalls zwischengespeichert worden ist, werden im weiteren erneut mit Hilfe der vorzugsweise als Additionsmittel ausgebildeten Kombinationsmittel KM des Programmmoduls PGM kombiniert bzw. addiert. Infolge dieser Operation entsteht erneut das modifizierte Audiosignal MAS, das vorzugsweise erneut zur Verbesserung der Signalqualität mit dem Präsenzfilter PRF gefiltert wird. Es ist aber auch erneut möglich, dass das modifizierte Audiosignal MAS, wie bei der Beschreibung der FIGUR 2 erläutert, ohne weitere Filterung durch das Präsenzfilter PRF dem Filter FIL zugeführt wird.The calculation means MBVF then supply from these two input variables the amplification factor VF with which the amplifier VS can be controlled. Thus, at the output of the amplifier VS, the band-pass filtered audio signal VSFK amplified by the amplification factor VF is applied again. This reinforced Bandpass-filtered audio signal VSFK and the audio signal AS, which may have been temporarily stored, are subsequently combined or added again with the aid of the combination means KM of the program module PGM, which are preferably designed as addition means. As a result of this operation, the modified audio signal MAS again arises, which is preferably filtered again to improve the signal quality with the presence filter PRF. However, it is also possible again for the modified audio signal MAS, as explained in the description of FIG. 2, to be supplied to the filter FIL without further filtering by the presence filter PRF.

FIGUR 7 zeigt eine Realisierungsform des Steuerungsvorrichtung STV gemäß der FIGUR 3. Das Audiosignal AS wird zur Isolation der ersten Frequenzkomponente FK mit einem als Hardware-Baustein ausgebildeten Bandpassfilter BPF1 bandpassgefiltert und zur Isolation der zweiten Frequenzkomponente FK' mit einem als Hardware-Baustein ausgebildeten Tiefpassfilter TPF1 tiefpassgefiltert. Während die erste Frequenzkomponente FK verstärkt wird, wird mit der zweiten Frequenzkomponente FK' der die Verstärkung der ersten Frequenzkomponente FK bestimmender Verstärkungsfaktor VF erzeugt.FIG. 7 shows an embodiment of the control device STV according to FIG. 3. The audio signal AS is band-pass filtered to isolate the first frequency component FK with a bandpass filter BPF1 designed as a hardware component and to isolate the second frequency component FK 'with a lowpass filter TPF1 designed as a hardware component low-pass filtered. While the first frequency component FK is amplified, the amplification factor VF determining the gain of the first frequency component FK is generated with the second frequency component FK '.

Anstelle des Tiefpassfilters TPF1 kann alternativ auch ein weiteres als Hardware-Baustein ausgebildetes Bandpassfilter oder sogar das die erste Frequenzkomponente FK erzeugende Bandpassfilter BPF1 verwendet werden. Im letztgenannten Fall wären die beiden Frequenzkomponenten FK, FK' gleich (FK=FK').Instead of the low-pass filter TPF1, alternatively, another bandpass filter designed as a hardware component or even the bandpass filter BPF1 generating the first frequency component FK may be used. In the latter case, the two frequency components FK, FK 'would be the same (FK = FK').

Das bandpassgefilterte Audiosignal FK bzw. die mit dem Bandpassfilter BPF1 isolierte Frequenzkomponente FK wird zur deren Verstärkung an den Eingang eines mit dem Verstärkungsfaktor VF steuerbaren als Hardware-Baustein ausgebildeten Verstärker VS1 gelegt. Für die Ermittlung des Verstärkungsfaktor VF sind in der Steuerungsvorrichtung STV als Hardware-Baustein ausgebildete Mittel zur Berechnung von Signaleinhüllende und/oder Signalenergie MBSE1 vorhanden, die vorzugsweise aus der Serienschaltung von einem Gleichrichter GLR und einem weiteren Tiefpassfilter TPF2 bestehen und die aus dem tiefpassgefilterten Audiosignal FK' eine Eingangsgröße für ebenfalls als Hardware-Baustein ausgebildete Mittel zur Berechnung des Verstärkungsfaktors MBVF1 der Steuerungsvorrichtung STV liefern. Die Berechnungsmittel MBVF1 liefern dann den Verstärkungsfaktor VF, mit dem der Verstärker VS1 steuerbar ist. Am Ausgang des Verstärkers VS1 liegt somit ein mit dem Verstärkungsfaktor VF verstärktes bandpassgefiltertes Audiosignal VSFK an. Dieses verstärkte bandpassgefilterte Audiosignal VSFK und das Audiosignal AS werden im weiteren mit Hilfe von vorzugsweise als Additionsmittel und als Hardware-Baustein ausgebildeten Kombinationsmittel KM1 der Steuerungsvorrichtung STV kombiniert bzw. addiert. Infolge dieser Operation entsteht das modifizierte Audiosignal MAS, das vorzugsweise zur Verbesserung der Signalqualität mit einem als Hardware-Baustein ausgebildeten Präsenzfilter PRF1 gefiltert wird. Es ist aber auch möglich, dass das modifizierte Audiosignal MAS, wie bei der Beschreibung der FIGUR 3 erläutert, ohne weitere Filterung durch das Präsenzfilter PRF dem Endverstärker EVS zugeführt wird.The band-pass filtered audio signal FK or the frequency component FK isolated with the bandpass filter BPF1 is applied to amplify it to the input of an amplifier VS1 which can be controlled by the amplification factor VF and is designed as a hardware component. For determining the amplification factor VF, means formed in the control device STV as a hardware module for calculating the signal envelope and / or the signal energy MBSE1 are preferably present consist of the series circuit of a rectifier GLR and another low-pass filter TPF2 and provide from the low-pass filtered audio signal FK 'an input for also designed as a hardware module means for calculating the gain MBVF1 the control device STV. The calculating means MBVF1 then supply the amplification factor VF with which the amplifier VS1 can be controlled. Thus, at the output of the amplifier VS1 there is a band-pass filtered audio signal VSFK amplified by the amplification factor VF. This amplified bandpass filtered audio signal VSFK and the audio signal AS are further combined or added with the aid of combination means KM1 of the control device STV, which are preferably designed as an addition means and as a hardware component. As a result of this operation, the modified audio signal MAS is produced, which is preferably filtered to improve the signal quality with a presence filter PRF1 embodied as a hardware component. But it is also possible that the modified audio signal MAS, as explained in the description of FIGURE 3, without further filtering by the presence filter PRF is supplied to the power amplifier EVS.

Claims (11)

  1. Method for controlling the bass reproduction of audio signals in electroacoustic converters, in which
    a) frequency components (FK, FK') of the audio signal (AS) are isolated and amplified (VS, VS1)with a gain factor (VF) calculated on the basis of the audio signal (AS),
    b) the amplified frequency components (VSFK) of the audio signal (AS) and the audio signal (AS) are combined (KM, KM1) such as to produce a modified audio signal (MAS),
    c) the modified audio signal (MAS) is fed to the electroacoustic converter (EAW),
    characterised in that
    d) the audio signal (AS) is bandpass filtered (BPF, BPF1) for isolation and amplification of first frequency components (FK),
    e) for calculation (MBVF, MBVF1) of the gain factor (VF)
    e1) the audio signal (AS) is lowpass and or bandpass filtered (BPF, BPF1, TPF, TPF1) for isolation of second frequency components (FK'),
    e2) the envelope and/or the energy of the unfiltered, lowpass filtered and/or bandpass filtered audio signal (AS, FK') is calculated (MBSE, MBSE1).
  2. Method according to claim 1, characterized in that the bandpass filtering is executed with a "Finite Impulse Response" filter (FIR-F).
  3. Method according to claim 1, characterized in that the bandpass filtering is executed with an "Infinite Impulse Response" filter (IIR-F).
  4. Method according to claim 2, characterized in that the audio signal (AS) to be combined with the amplified frequency components (VFK) is buffered (ZWS).
  5. Method according to one of the claims 1 to 4, characterized in that bandpass filtering for isolation and amplification of the frequency components and for calculation of the gain factor is undertaken with a single bandpass filter (BPF, BPF1).
  6. Method according to one of the claims 1 to 4, characterized in that the bandpass filtering for the isolation and amplification of the frequency components is undertaken with one bandpass filter (BPF, BPF1) and bandpass filtering for calculation of the gain factor is undertaken with a further bandpass filter.
  7. Method according to one of the claims 1 to 6, characterized in that the modified audio signal (MAS) is filtered (PRF, PRF1) for amplification of selected frequencies.
  8. Method according to one of the claims 1 to 7, characterized in that the method runs in an electronic device for output and/or reproduction of audio signals.
  9. Device for controlling the bass reproduction in electroacoustic converters, in which
    (a) isolation means (BPF, BPF1, TPF, TPFL) at the input of which the audio signal (AS) is present isolate the frequency components (FK, FK') of the audio signal (AS),
    (b) calculation means (MBVF, MBVF1) are present which calculate a gain factor (VF) based on the audio signal (AS),
    (c) an amplifier (VS, VS1) is present which is connected to the isolation and calculation means in such a way that the frequency components (FK, FK') of the audio signal (AS) are amplified with the gain factor (VF) calculated,
    (d) combination means (KM, KM1) are present, at the input of which the audio signal (AS) and the amplified frequency components (VSFK) of the audio signal (AS) are present and which combine the audio signal (AS) and the amplified frequency components (VSFK) of the audio signal (AS) in such a way that at the output of the combination means (KM, KM1) a modified audio signal (MAS) intended for the electroacoustic converter (EAW) is present.
    characterised in that
    (e) at least one bandpass filter (BPF, BPF1) or at least one bandpass filter (BPF, BPF1) and lowpass filter (TPF, TPF1) in each case is present for isolation of a first frequency component (FK) and a second frequency component (FK') of the audio signal (AS),
    (f) of the bandpass filters (BPF, BPF1) one bandpass filter is connected on its output side with the amplifiers (VS, VS1) for isolation of the first frequency component (FK)
    (g) means are present for calculating signal envelopes and/or signal energy (MBSE, MBSE1), at which on the input side the unfiltered, lowpass-filtered and/or bandpass-filtered audio signal (AS, FK') is present,
    (h) the calculation means (MBVF, MBVF1) for calculating the gain factor (VF) is connected on its input side with the means for calculating the signal envelope and/or signal energy (MBSE, MBSE1) and on its output side with the amplifier (VS, VS1) for setting the gain factor (VF).
  10. Device according to claim 9, characterized in that a presence filter (PRF, PRF1) is available for amplifying selected frequencies of the modified audio signal (MAS).
  11. Device according to claim 9 or 10, characterized in that the device is integrated or contained in an electronic device for output and/or reproduction of audio signals.
EP01980187A 2001-09-21 2001-09-21 Method and device for controlling the bass reproduction of audio signals in electroacoustic transducers Expired - Lifetime EP1428411B2 (en)

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PCT/DE2001/003653 WO2003028405A1 (en) 2001-09-21 2001-09-21 Method and device for controlling the bass reproduction of audio signals in electroacoustic transducers

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WO2003028405A1 (en) 2003-04-03
EP1428411B2 (en) 2011-11-30
CN1550121A (en) 2004-11-24
US20050002534A1 (en) 2005-01-06
HK1069705A1 (en) 2005-05-27
EP1428411A1 (en) 2004-06-16
DE50112650D1 (en) 2007-08-02
US7574009B2 (en) 2009-08-11

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