EP1428411A1 - Procede et dispositif de commande de la restitution des basses de signaux audio dans des transducteurs electroacoustiques - Google Patents

Procede et dispositif de commande de la restitution des basses de signaux audio dans des transducteurs electroacoustiques

Info

Publication number
EP1428411A1
EP1428411A1 EP01980187A EP01980187A EP1428411A1 EP 1428411 A1 EP1428411 A1 EP 1428411A1 EP 01980187 A EP01980187 A EP 01980187A EP 01980187 A EP01980187 A EP 01980187A EP 1428411 A1 EP1428411 A1 EP 1428411A1
Authority
EP
European Patent Office
Prior art keywords
audio signal
bandpass
filter
frequency components
bpf
Prior art date
Legal status (The legal status is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the status listed.)
Granted
Application number
EP01980187A
Other languages
German (de)
English (en)
Other versions
EP1428411B2 (fr
EP1428411B1 (fr
Inventor
Roland Aubauer
Stefano Ambrosius Klinke
Current Assignee (The listed assignees may be inaccurate. Google has not performed a legal analysis and makes no representation or warranty as to the accuracy of the list.)
Gigaset Communications GmbH
Original Assignee
Siemens AG
Priority date (The priority date is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the date listed.)
Filing date
Publication date
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Application filed by Siemens AG filed Critical Siemens AG
Publication of EP1428411A1 publication Critical patent/EP1428411A1/fr
Application granted granted Critical
Publication of EP1428411B1 publication Critical patent/EP1428411B1/fr
Publication of EP1428411B2 publication Critical patent/EP1428411B2/fr
Anticipated expiration legal-status Critical
Expired - Lifetime legal-status Critical Current

Links

Classifications

    • 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

  • the invention relates to a method for controlling the bass reproduction of audio signals in electroacoustic transducers according to the preamble of patent claim 1 and a device for controlling the bass reproduction of audio signals in electroacoustic transducers according to the preamble of patent claim 9.
  • the bass reproduction of audio signals in an electroacoustic transducer is dependent on the size of the electroacoustic transducer, the loudspeaker or the earpiece.
  • FIGURE 1 shows a typical frequency response of a small loudspeaker.
  • Electronic audio devices in which such small electroacoustic transducers are used and in which bass reproduction is consequently unsatisfactory are primarily audio devices (devices for outputting and / or reproducing audio signals) of communication and information technology as well as entertainment and consumer goods electronics, such as Mobile and cordless telephone handsets, notebooks, personal digital assistants, mini radios, clock radios, portable music players, etc.
  • a known psychoacoustic principle can be used to improve bass reproduction with a small loudspeaker. This principle is referred to as “residual hearing (hearing of missing fundamentals)” or as a “virtual pitch”. 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 corresponding combination of its harmonics.
  • a method based on the psychoacoustic principle is known from WO 00/15003, in which the harmonics present in the audio signal are amplified.
  • low frequency components of the audio signal are isolated to form a low-frequency audio signal, the isolated deep frequency components are filtered with a large number of bandpass filters, and the bandpass-filtered frequency components in an amplifier which can be controlled with respect to the gain factor amplified, the gain factor being obtained from the envelope of the band-pass filtered frequency components, and producing a simulated low-frequency audio signal by combining the original audio signal with the amplified frequency components.
  • the object on which the invention is based is to control the bass reproduction of audio signals in electroacoustic transducers based on the psychoacoustic principle referred to as “virtual pitch” or “residual hearing (hearing of missing fundamental) ⁇ ” in such a way that the perception of the virtual bass reproduction the audio signals is improved over the prior art.
  • This object is achieved both on the basis of the method defined in the preamble of patent claim 1 by the features specified in the characterizing part of patent claim 1 and on the basis of the device defined in the preamble of patent claim 9 by the features specified in the characterizing part of patent claim 9.
  • the idea that makes up the invention consists in controlling the reproductions of the low frequencies or basses emitted in the electroacoustic transducer by amplifying the harmonic harmonics already contained in the audio signal in such a way that the listener feels improved bass reproduction or perceives.
  • the control or simulation can be both digital (claim 1), by a program module in the digital signal processor DSP of the electronic device for output and / or reproduction of audio signals with the electroacoustic transducer, and analog (claim 9), by a Hardware connection between the digital / analog converter and the final amplifier of the electronic device for output and / or playback of audio signals with the electroacoustic converter.
  • the program module and the hardware circuit only the harmonic waves which are above the resonance frequency of the electroacoustic transducer, in particular the loudspeaker, are amplified in order to simulate the perception of the fundamental frequency.
  • the extraction or isolation of the harmonics is achieved in the program module by means of bandpass filtering and in the hardware circuit by means of a bandpass filter, while the amplification of the harmonics is controlled by a software-based gain factor in the program module and in the hardware circuit in a suitably designed one Gain Controlled Amplifier runs.
  • the gain factor is preferably determined by frequency Components of the audio signal controlled below the resonance frequency or cutoff frequency of the electroacoustic transducer.
  • the advantage of the method according to claim 1 is 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. This particularly avoids distortion of the audio signal.
  • the method according to the invention places lower demands on the computing power and the memory requirement in the digital signal processor.
  • the modified audio signal is filtered to amplify selected frequencies.
  • FIGS. 2 to 7. Two exemplary embodiments of the invention are explained with reference to FIGS. 2 to 7. Show it:
  • FIGURE 2 the digital implementation of the method according to the invention in the form of a program module in a digital Signal processor of an electronic radio device for the output and / or playback of 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 output and / or playback of audio signals
  • FIGURE 4 shows a first form of implementation of the program module according to FIGURE 2
  • FIG. 5 shows a second form of implementation of the program module according to FIG. 2,
  • FIG. 6 shows a third form of implementation of the program module according to FIG. 2,
  • FIG. 7 shows a form of implementation of the control device according to FIG. 3.
  • FIGURE 2 shows, as a second exemplary embodiment, in the form of a functional or block diagram, the speech processing path in a radio device FG for the output and / or reproduction of audio signals, in particular voice signals, in which the invention is implemented in a program module PGM of a digital signal processor DSP (digital implementation).
  • the radio FG receives an analog radio signal FS via an antenna ANT, on which encoded speech 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 fed 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 electro-acoustic transducers PGM of the digital signal processor DSP supplied.
  • 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 passed to a digital-to-analog converter DAW and then amplified in a power amplifier EVS before the speech information contained in the modified audio signal MAS is output by an electroacoustic converter EAS, which is preferably designed as a loudspeaker ,
  • FIG. 3 shows, as a second exemplary embodiment in the form of a functional or block diagram, the voice processing path in the radio device FG, in which, in contrast to FIG. 2, the invention outside the digital signal processor DSP in the analog part of the radio device FG in a device for controlling the bass reproduction of audio signals in electroacoustic transducers STV is implemented (analog implementation).
  • the voice signal processing in the radio device FG in turn begins with the fact that the analog radio signal FS, on which encoded speech information is modulated, is fed to the receiver EMP via the antenna ANT.
  • the microprocessor MP and the analog-digital converter ADW in turn, generate the digital demodulated signal DDS 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 obtained again 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 is converted accordingly in the digital-to-analog converter DAW.
  • the converted audio signal AS is then the device for controlling the bass reproduction of audio signals in electroacoustic transducers STV supplied, where a modified audio signal MAS is generated from the audio signal AS.
  • the modified audio signal MAS is then amplified in the power amplifier EVS before the speech 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 form of implementation of the program module PGM according to FIG. 2.
  • the audio signal AS is bandpass-filtered for isolation of a first frequency component FK with a bandpass filter BPF implemented with software and for isolation of a second frequency component FK X with a lowpass filter realized with software TPF , While the first frequency component FK is being amplified, an amplification factor VF determining the amplification of the first frequency component FK is generated with the second frequency component FK X.
  • a further bandpass filter implemented by software or even the bandpass filter BPF generating the first frequency component FK can alternatively be used.
  • the bandpass filter BPF is preferably designed as a finite impulse response filter (FIR filter) FIR-F or alternatively as an "infinite impulse response ⁇ filter (IIR filter) IIR-F. If the bandpass filter BPF is a finite impulse response" Designed filter FIR-F, the program module PGM contains a buffer ZWS for buffering the audio signal AS. 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 FIG. 4, the buffer ZWS is shown as a dashed block.
  • the bandpass-filtered audio signal FK or the frequency component FK isolated with the bandpass filter BPF is applied to amplify it at the input of an amplifier VS which can be controlled with the amplification factor VF and is implemented using software.
  • the program module PGM contains software-implemented means for calculating signal envelopes and / or signal energy MBSE, which supply an input variable from the low-pass-filtered audio signal FK ⁇ for means also used for software to calculate the gain factor MBVF of the program module PGM , The calculation means MBVF then deliver the gain factor VF with which the amplifier VS can be controlled.
  • a bandpass-filtered audio signal VSFK is thus present at the output of the amplifier VS with the amplification factor VF.
  • This operation results in the modified audio signal MAS, which is preferably filtered to improve the signal quality with a presence filter PRF implemented by software.
  • the modified audio signal MAS is fed to the filter FIL without further filtering by the presence filter PRF.
  • FIGURE 5 shows, starting from FIGURE 4, a second implementation of the program module PGM according to FIGURE 2.
  • the audio signal AS is bandpass filtered again with the bandpass filter BPF for isolation of the first frequency component FK and lowpass filtered for isolation of the second frequency component FK with the lowpass filter TPF. While the first frequency component FK is amplified again, the second frequency component FK again amplifies the Gain factor VF determines first frequency component FK.
  • the bandpass filter BPF is again preferably designed as a finite impulse response filter (FIR filter) FIR-F or alternatively as an "infinite pulse response 1 " filter (IIR filter) IIR-F. If the bandpass filter BPF is designed as a finite impulse response filter FIR-F, the program module PGM again contains the buffer ZWS for buffering the audio signal AS. This buffer ZWS is again when the bandpass filter BPF as an infinite impulse response " 1 - Filter IIR-F is formed, not required. In order to illustrate this in FIGURE 5, the buffer ZWS is shown as a dashed block.
  • the bandpass-filtered audio signal FK or the frequency component FK isolated with the bandpass filter BPF is applied to the input of an amplifier VS controllable with the gain factor VF, as in FIG.
  • the program module PGM again contains the means for calculating the signal envelope and / or signal energy MBSE, which again provide an input variable from the low-pass filtered audio signal FK ⁇ for the means for calculating the gain factor MBVF of the program module PGM.
  • the calculation means MBVF is supplied with a further input variable which comes from further means for calculating signal envelopes and / or signal energy MBSE.
  • the further input variable is calculated by the calculation means MBSE from the unfiltered audio signal AS.
  • the calculation means MBVF then deliver the gain factor VF from these two input variables, with which the amplifier VS can be controlled again.
  • the bandpass-filtered audio signal VSFK is thus present again at the output of the amplifier VS with the amplification factor VF.
  • This amplified bandpass-filtered audio signal VSFK and the audio signal AS which may have been buffered, are subsequently combined or added again with the aid of the combination means KM of the program module PGM, which are preferably again designed as addition means.
  • the modified audio signal MAS is produced, which is preferably filtered again with the presence filter PRF to improve the signal quality.
  • the modified audio signal MAS is fed to the filter FIL without further filtering by the presence filter PRF.
  • FIGURE 6 is based on FIG 4 shows a third form of implementation of the program module PGM in accordance with FIG 2.
  • the audio signal AS is again band-pass filtered to isolate the first frequency component FK with the band-pass filter BPF and again low-pass filtered to isolate the second frequency component FK ⁇ with the low pass filter LPF. While the first frequency component FK is amplified again, the amplification factor VF which determines 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 filter (FIR filter) FIR-F or alternatively as an "infinite impulse response" filter (IIR filter) IIR-F. If the bandpass filter BPF is designed as a finite impulse response "filter FIR-F, the program module PGM again contains the buffer ZWS for buffering the audio signal AS. This buffer ZWS is again when the bandpass filter BPF as an infinite impulse response" filter IIR-F is not required. In order to show this in FIGURE 6, the buffer ZWS is shown as a dashed block.
  • the bandpass-filtered audio signal FK or the frequency component FK isolated with the bandpass filter BPF is, as in FIGS. 4 and 5, applied to the input of the amplifier VS which can be controlled with the amplification factor VF in order to amplify it.
  • the program module PGM again contains the means for calculating signal envelope and / or signal energy MBSE, which provide an input variable from the low-pass filtered audio signal FK ⁇ for means for calculating the gain factor MBVF of the program module PGM.
  • the calculation means MBVF is supplied with a further input variable which comes from further means for calculating signal envelopes and / or signal energy MBSE.
  • the further input variable-e- is calculated by the calculation means MBSE from the bandpass-filtered audio signal FK.
  • the calculation means MBVF then deliver the gain factor VF from these two input variables, with which the amplifier VS can be controlled.
  • the bandpass-filtered audio signal VSFK which is amplified with the amplification factor VF, is thus again present at the output of the amplifier VS.
  • This reinforced bandpass-filtered audio signal VSFK and the audio signal AS which may have been buffered, are subsequently combined or added again using the combination means KM of the program module PGM, which are preferably designed as addition means.
  • the modified audio signal MAS is generated again, which is preferably filtered again with the presence filter PRF to improve the signal quality.
  • the modified audio signal MAS is fed to the filter FIL without further filtering by the presence filter PRF.
  • FIG. 7 shows a form of implementation of the control device STV according to FIG. 3.
  • the audio signal AS is bandpass-filtered for isolation of the first frequency component FK with a bandpass filter BPF1 designed as a hardware module and for isolation of the second frequency component FK with a hardware module trained low-pass filter TPF1 low-pass filtered. While the first frequency component FK is amplified, the second frequency component
  • the bandpass-filtered audio signal FK or the frequency component FK isolated with the bandpass filter BPF1 is placed at the input of an amplifier VS1 designed as a hardware component that can be controlled with the amplification factor VF in order to amplify it.
  • means for calculating signal envelopes and / or signal energy MBSE1 are provided in the control device STV as hardware modules, which are preferably They consist of a series connection of a rectifier GLR and a further low-pass filter TPF2 and which supply an input variable from the low-pass filtered audio signal FK X for means for calculating the gain factor MBVFl of the control device STV, which are also designed as hardware components.
  • the calculation means MBVF1 then deliver the gain factor VF with which the amplifier VS1 can be controlled.
  • a bandpass-filtered audio signal VSFK is thus present at the output of the amplifier VS1, amplified with the amplification factor VF.
  • This amplified bandpass-filtered audio signal VSFK and the audio signal AS are further combined or added using combination means KM1 of the control device STV, which are preferably designed as addition means and as a hardware component.
  • the modified audio signal MAS is produced, which is preferably filtered with a presence filter PRF1 designed as a hardware module in order to improve the signal quality.
  • the modified audio signal MAS as explained in the description of FIG. 3, to be fed to the power amplifier EVS without further filtering by the presence filter PRF.

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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)

Abstract

La présente invention concerne la commande de la restitution des basses de signaux audio (AS) dans des transducteurs électroacoustiques (EAW) sur la base du principe psychoacoustique désigné sous le terme de « hauteur virtuelle » ou d' « audition résiduelle (audition de la fondamentale virtuelle) » de manière telle que la perception de la restitution virtuelle des basses des signaux audio (AS) soit améliorée par rapport à l'état actuel de la technique. A cet effet, la restitution des fréquences faibles ou basses introduites dans le transducteur électroacoustique (EAW) est commandée par l'amplification des ondes harmoniques déjà contenues dans le signal audio (AS), dans le sens d'une simulation, de façon que l'auditeur ressente ou perçoive une meilleure restitution des basses. La commande ou simulation peut alors être tant numérique (revendication 1) grâce à un module de programme (PGM) situé dans un processeur de signaux numériques (DSP) d'un appareil électronique pour la production et / ou la restitution de signaux audio (AS) à l'aide du transducteur électroacoustique (EAW), qu'analogique (revendication 9) grâce à une connexion matérielle entre un convertisseur numérique / analogique (DAW) et un amplificateur final (EVS) de l'appareil électronique (FG) pour la production et / ou la restitution de signaux audio (AS) à l'aide du transducteur électroacoustique (EAW).
EP01980187A 2001-09-21 2001-09-21 Procede et dispositif de commande de la restitution des basses de signaux audio dans des transducteurs electroacoustiques Expired - Lifetime EP1428411B2 (fr)

Applications Claiming Priority (1)

Application Number Priority Date Filing Date Title
PCT/DE2001/003653 WO2003028405A1 (fr) 2001-09-21 2001-09-21 Procede et dispositif de commande de la restitution des basses de signaux audio dans des transducteurs electroacoustiques

Publications (3)

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EP1428411A1 true EP1428411A1 (fr) 2004-06-16
EP1428411B1 EP1428411B1 (fr) 2007-06-20
EP1428411B2 EP1428411B2 (fr) 2011-11-30

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US (1) US7574009B2 (fr)
EP (1) EP1428411B2 (fr)
CN (1) CN1274184C (fr)
DE (1) DE50112650D1 (fr)
HK (1) HK1069705A1 (fr)
WO (1) WO2003028405A1 (fr)

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Also Published As

Publication number Publication date
DE50112650D1 (de) 2007-08-02
US7574009B2 (en) 2009-08-11
CN1274184C (zh) 2006-09-06
EP1428411B2 (fr) 2011-11-30
HK1069705A1 (en) 2005-05-27
US20050002534A1 (en) 2005-01-06
EP1428411B1 (fr) 2007-06-20
CN1550121A (zh) 2004-11-24
WO2003028405A1 (fr) 2003-04-03

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