EP1130577B1 - Procédé de reconstruction des basses fréquences du signal de parole à partir de fréquences moyennes - Google Patents

Procédé de reconstruction des basses fréquences du signal de parole à partir de fréquences moyennes Download PDF

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Publication number
EP1130577B1
EP1130577B1 EP01102129A EP01102129A EP1130577B1 EP 1130577 B1 EP1130577 B1 EP 1130577B1 EP 01102129 A EP01102129 A EP 01102129A EP 01102129 A EP01102129 A EP 01102129A EP 1130577 B1 EP1130577 B1 EP 1130577B1
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Prior art keywords
frequency
speech signal
signal
speech
fundamental
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EP01102129A
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German (de)
English (en)
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EP1130577A3 (fr
EP1130577A2 (fr
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Jürgen Schultz
Klaus Dr. Schaaf
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Volkswagen AG
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Volkswagen AG
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    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L21/00—Speech or voice signal processing techniques to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility
    • G10L21/02—Speech enhancement, e.g. noise reduction or echo cancellation
    • G10L21/038—Speech enhancement, e.g. noise reduction or echo cancellation using band spreading techniques

Definitions

  • the invention relates to a method and an apparatus for the reconstruction of low-frequency speech components from medium-high frequency components.
  • the signal is improved in that either noise components are filtered out or very strongly disturbed frequency range are completely filtered out of the signal.
  • US Pat. No. 5,842,160 A discloses a method for improving the quality of a digital voice transmission in which different data volumes are assigned to different frequency bands depending on the energy content.
  • the nature of the coding and transmission results in low-energy signal areas, which lead to gaps in the received signal spectrum. These gaps are filled by signals synthesized from the existing data so that a more natural sounding speech signal is achieved.
  • the method described above and the associated devices is based on the disadvantage that the speech signal is not reconstructed at all or only in an inadequate form in order to produce the most natural possible source speech signal.
  • DVE digital voice enhancement
  • two microphones are mounted above each row of seats in a motor vehicle, so that it is, for example. All vehicle occupants is allowed to participate in a telephone conversation.
  • the system transmits the voice recorded at the front of the microphone to the rear standard loudspeakers and vice versa.
  • the system is thus fully connected to the handsfree telephone and the radio / CD / navigation device. It significantly improves the communication within the vehicle, especially when driving fast.
  • the level of the vehicle interior noise increases very strongly to low frequencies, so that the language is covered there by the noise.
  • all frequencies are cut off below, for example, 200 to 500 Hz, depending on the speed.
  • the result is that the speech fundamental frequency and the first multiples (harmonics) in the transmitted signal are missing.
  • the language thus sounds like a telephone, as typically a telephone network allows a sound transmission only above 350 Hz.
  • the invention is therefore based on the technical problem of further developing the known from the prior art method and the associated apparatus for the reconstruction of low-frequency speech components of medium frequency components and to design that for a reproduction of the disturbed speech signal as natural as possible reproduction possible.
  • the above-indicated technical problem is solved by a method having the features of claim 1. First, at least two adjacently arranged frequency components with an increased amplitude in the voice signal are determined above a cutoff frequency. Thereafter, the fundamental frequency of the speech signal is determined as a frequency difference between the at least two adjacent frequency components. Finally, the low-frequency frequency range below the cut-off frequency is reconstructed with the aid of the determined fundamental frequency and the speech signal. The thus generated synthetic speech signal can then be output directly via a playback device or stored for later transmission.
  • low-frequency signal components of the speech signal are generated synthetically, that is to say reconstructed, and admixed with the remaining recorded speech signal.
  • the reconstruction of the low-frequency speech components is done on the basis of the non-filtered speech signals. This is exploited that the low-frequency speech components are accompanied by higher-frequency components of the harmonics, so that can be estimated from the remaining signal, the missing portions.
  • the frequencies of the harmonics of the fundamental frequency arranged below the limit frequency are preferably determined and used in addition to the fundamental frequency for a reconstruction of the low-frequency frequency range.
  • the maximum information regarding the undisturbed speech signal is utilized from the spectrally evaluated section of the speech signal.
  • the frequencies used for the reconstruction are combined with a respective spectral distribution and a predetermined amplitude to form a synthetic spectrum which corresponds to the frequency range below the cutoff frequency in the speech signal. From this frequency section and the speech signal above the cutoff frequency, the reconstructed speech signal is then composed.
  • the low-frequency speech component thus no longer has a noise signal, since it is composed exclusively of frequency components of the speech signal.
  • the low-frequency speech component can also be determined directly from the speech signal.
  • a comb filter consisting of several band filters is set up on the basis of the fundamental frequency and the frequencies of the harmonics arranged below the cutoff frequency, the frequency positions of the individual bandpass filters corresponding to the cutoff frequencies and the harmonics.
  • the speech signal is then filtered in the range below the cutoff frequency, whereby the signal components which belong to the actual speech signal are transmitted. Also in this way, a reconstruction of a largely undisturbed speech signal in the low-frequency range of the speech signal is possible.
  • the amplitude of the at least one frequency signal generated below the cutoff frequency is determined as a function of the amplitudes of the frequency signals analyzed above the cutoff frequency.
  • typical amplitude profiles of speech signals can be used in order to achieve as exact as possible adaptation to a natural speech signal not only in the frequency components but also in the amplitude distribution of the frequency components.
  • the cutoff frequency is determined as a function of the noise level, that is to say, in particular, on the size of the interfering signal.
  • the cut-off frequency can also be determined as a function of the driving speed.
  • a development consists in that the speech signal is subjected to a noise suppression before conversion.
  • the conventional methods known from the prior art can be used to perform a pretreatment of the speech signal.
  • the speech components then emerge more clearly in the spectrum and can be recognized more clearly and therefore more accurately and reconstructed.
  • One application of the method described above is to reproduce voice signals recorded in a moving motor vehicle in order to reproduce the most natural possible language impression.
  • Another application of the method according to the invention is to reproduce a voice signal transmitted by means of a telephone connection.
  • the underlying problem is that the voice signals for telephone connections in the frequency range below 350 Hz contain no information. Therefore, for a faithful reproduction of the speech signal, the low-frequency speech component must be reconstructed from the frequency range above 350 Hz. This can be carried out in a particularly advantageous manner by the method according to the invention.
  • Fig. 1 shows a frequency-amplitude diagram of the interior noise level in a moving motor vehicle for different speeds between 60 Km / h and 160 Km / h.
  • the inner noise level rises sharply in comparison to the other frequencies of the inner noise signal.
  • a determination, so filtering out the speech signal from the interior noise signal is considerably more difficult.
  • Fig. 2 shows a speech signal superimposed on a background signal in a time-frequency representation as a spectrogram.
  • This spectrogram is obtained, for example, by a Fourier transform (FFT) from a microphone signal.
  • FFT Fourier transform
  • different gray levels of the individual segments of the spectrogram indicate different intensities.
  • narrow-band frequency components that run largely parallel to each other over short periods of time. These latter narrow-band frequency components represent harmonics of the fundamental frequency of the corresponding voice signal, which are evaluated according to the invention as described below.
  • FIG. 3 shows a spectrogram of the speech signal shown in FIG. 2 without the background noise, so that the low-frequency speech components can also be recognized as narrow-band frequency components in the spectrogram below 500 Hz. These language parts need to be reconstructed.
  • FIG. 4 further shows the previously described speech signal, in which the speech components are cut off below a cutoff frequency of approximately 400 Hz. Such a signal is approximately the same as the voice signal transmitted on a telephone connection.
  • FIG. 5 shows an example of a reconstructed speech signal in the range below the cutoff frequency of approximately 400 Hz
  • FIG. 6 shows the composite reconstructed speech signal from the reconstructed speech component shown in FIG. 5 and the frequency component shown in FIG. 4 above the cutoff frequency of the original one spectrum. How the reconstructed speech parts are obtained will be described in detail below with reference to FIGS. 7 to 9.
  • FIG. 7 shows in a block diagram an apparatus for the reconstruction of low-frequency speech components from medium-high frequency components.
  • the speech signal is fed to a means 4 for determining frequency components ⁇ fa1 , ⁇ fa2 ,... Of maxima in the speech signal above a predetermined limit frequency ⁇ 0 .
  • the speech signal is first passed through a bandpass filter 6, so that only the frequency components between the cutoff frequency ⁇ 0 and another frequency ⁇ 1 cut out and forwarded to further processing.
  • ⁇ 0 is, for example, in the range of 200 to 500 Hz, in particular 350 Hz
  • the frequency ⁇ 1 is, for example, in the range of 800 Hz.
  • the thus-filtered frequency portion of the speech signal is mixed in the mixing element 8, so that the sum and difference frequencies of the frequency components contained in the cut-out portion of the speech signal are formed.
  • the difference frequencies are the difference frequencies, so that the signal emerging from the mixing element 8 is processed by means of a low-pass filter, so that only frequency components below an adjustable frequency ⁇ 2 are transmitted.
  • the smallest difference frequency can be determined, the distance between two in the Speech signal adjacent to each other arranged spectral components corresponds. Since these are two harmonics of the fundamental frequency, the difference frequency represents the fundamental frequency ⁇ g .
  • This fundamental frequency is then fed to means 12 for reconstructing the speech signal. Via a further input of the means 12, the speech signal via a delay stage 14 and a low-pass filter 16 is supplied.
  • both the value of the fundamental frequency ⁇ g and a predetermined frequency section of the speech signal are available to the means 12 for reconstruction of the signal containing the speech.
  • the delay stage 14 serves to compensate for the time .DELTA.t, which is needed for the determination of the fundamental frequency ⁇ g and the low-pass filter 16 is a useful reduction of the amount of data, which is fed to the means 12 for the reconstruction of the speech signal.
  • the means 12 for the reconstruction of the speech signal below the cut-off frequency ⁇ 0 has two alternatives of procedures in terms of circuitry.
  • the fundamental frequency ⁇ g is used to generate a signal in the reconstructed speech signal corresponding to the root of the speech.
  • the aim is to generate all harmonics in the frequency section of the speech signal to be reconstructed, that is to simulate them.
  • the means 12 comprise a comb filter comprising a plurality of band filters whose spectral transmission functions are determined by the fundamental frequency ⁇ g and the frequencies ⁇ h1 , ⁇ h2 ,.. ,
  • the spectral transmission function of each bandpass filter is also defined over a predetermined width, so that corresponding spectral sections are filtered out of the speech signal in the range of low frequencies below the cutoff frequency ⁇ 0 . Since from the spectrogram only the proportions are filtered out, containing the speech signal, the speech signal is reconstructed from the spectrogram. If additionally a noise suppression is carried out, then the background noises are filtered out of the filtered-out signal components, so that an almost natural speech signal is generated.
  • the speech signal is delayed by a further delay stage 18 by a time difference ⁇ t, in order to make it possible to adapt to the time span necessary for the reconstruction of the low-frequency speech component.
  • a high-pass filter 20 in which the speech signal above the cut-off frequency ⁇ 0 is filtered out
  • both this high-pass filtered signal and the reconstructed speech signal for frequencies ⁇ ⁇ 0 converge in the summation element 22, from which the reconfigured spectrogram shown in FIG becomes.
  • This spectrogram therefore consists on the one hand of the frequency component reconstructed below the cutoff frequency ⁇ 0 and of the original frequency spectrum above the cutoff frequency ⁇ 0 .
  • the spectrogram thus produced, after conversion to a loudspeaker signal, results in almost natural-sounding speech reproduction.
  • the fundamental frequency ⁇ g in a speech signal does not remain constant due to the speech melody. Therefore, it is necessary to constantly redetermine the fundamental frequency ⁇ g . This can on the one hand be done by constantly going through the process described above, which has been previously described with reference to the elements 4, 6, 8 and 10. On the other hand, however, a more accurate adaptive tracking of the fundamental frequency ⁇ g can be performed. This is possible with a device which is shown in Fig. 8.
  • the fundamental frequency ⁇ g, 0 initially determined at the beginning of a speech signal is multiplied to N times the value by means of a multiplication element 24.
  • the (N-1) th harmonic is calculated to the fundamental frequency.
  • the frequency of these harmonics is hereinafter referred to as harmonic and the associated frequency denoted by ⁇ r .
  • the frequency ⁇ r is introduced via a Mehrtorschalter in a control loop.
  • the output of the multiplication element 24 is transferred from the multi-port switch 26 to the mixing element 28.
  • an estimated value ⁇ r new before and the Mehrtorschalter 26 is switched so that ⁇ r , new to the mixing element 28 is passed.
  • ⁇ r is exactly the frequency of the (N-1) th harmonic.
  • the mixing element 28 forms the difference between ⁇ r and ⁇ m .
  • a sine wave generator generates a sinusoidal signal with the frequency given by its input signal ⁇ d . This is fed to a mixing element 32 which mixes the speech signal and this sinusoidal signal. After mixing, the mixed signal is outputted from the mixing element 32, which is supplied to a control element 34 for detecting the frequency-dependent power distribution in the mixing signal with respect to the fixed frequency ⁇ m .
  • the power distribution will assume its maximum not at the frequency ⁇ m but at a position shifted by a difference value ⁇ .
  • a correction value to ⁇ can be determined, which is added to the current value of the frequency ⁇ r of the control harmonics added. This results in the new value of the frequency ⁇ r, new , which is fed again via the multiport switch 26 of the control loop. Subsequently, a mixture is again in the mixing element 28 with subsequent control sequence, as has been previously described.
  • the value ⁇ r is diverted from the control loop via a multiplication element 38 and output, in which the current frequency ⁇ r is applied with the factor 1 / N to adapt the value of the fundamental frequency ⁇ g, adapt .
  • the value of the fundamental frequency ⁇ g is constantly adaptively tracked, whereby the reconstruction of the low-frequency speech component from the medium-high frequency components is improved and brought closer to a natural speech signal.

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  • Engineering & Computer Science (AREA)
  • Acoustics & Sound (AREA)
  • Computational Linguistics (AREA)
  • Signal Processing (AREA)
  • Health & Medical Sciences (AREA)
  • Audiology, Speech & Language Pathology (AREA)
  • Human Computer Interaction (AREA)
  • Quality & Reliability (AREA)
  • Multimedia (AREA)
  • Physics & Mathematics (AREA)
  • Fittings On The Vehicle Exterior For Carrying Loads, And Devices For Holding Or Mounting Articles (AREA)
  • Compression, Expansion, Code Conversion, And Decoders (AREA)
  • Telephone Function (AREA)
  • Measurement Of Mechanical Vibrations Or Ultrasonic Waves (AREA)
  • Transmission Systems Not Characterized By The Medium Used For Transmission (AREA)
  • Mobile Radio Communication Systems (AREA)

Claims (16)

  1. Procédé de reconstruction de composantes vocales à basses fréquences à partir de composantes à moyenne fréquence,
    - dans lequel au moins deux composantes de fréquence (ωfa1, ωfa2, ...) voisines ayant une amplitude accrue sont déterminées dans le signal vocal au-dessus d'une fréquence limite (ω0) et
    - dans lequel la fréquence fondamentale (ωg) du signal vocal est déterminée sous la forme de la différence de fréquence entre lesdites au moins deux composantes de fréquence (ωfa1, ωfa2, ...) voisines et
    - dans lequel la plage des basses fréquences au-dessous de la fréquence limite (ω0) est reconstruite à l'aide de la fréquence fondamentale (ωg) déterminée et du signal vocal.
  2. Procédé selon la revendication 1, dans lequel les fréquences (ωh1, ωh2, ...) des harmoniques de la fréquence fondamentale (ωg) qui se trouvent au-dessous de la fréquence limite (ω0) sont déterminées à partir de la fréquence fondamentale (ωg) et sont utilisées en plus de la fréquence fondamentale, (ωg) pour la reconstruction de la plage des basses fréquences.
  3. Procédé selon la revendication 1, dans lequel les positions en fréquence des filtres passe-bande à l'aide desquels le signal vocal' est filtré dans la plage au-dessous de la fréquence limite (ω0) sont réglées à l'aide d'un filtre en peigne présentant plusieurs filtres passe-bande en se basant sur la fréquence fondamentale (ωg) et sur les fréquences des harmoniques qui se trouvent au-dessous de la fréquence limite (ω0).
  4. Procédé selon l'une des revendications 1 à 3, dans lequel la fréquence fondamentale (ωg) est déterminée à partir du signal vocal au début d'une section vocale contenant de la parole et la fréquence fondamentale (ωg) est ensuite asservie de manière adaptative.
  5. Procédé selon la revendication 4,
    - dans lequel, pour un asservissement adaptatif de la fréquence fondamentale (ωg), la fréquence (ωr) d'une harmonique standard est calculée en tant que Nième harmonique à partir de la valeur actuelle de la fréquence fondamentale (ωg),
    - dans lequel la différence entre la fréquence (ωr) de l'harmonique standard et une fréquence combinée fixe (ωm) est calculée,
    - dans lequel un signal sinusoïdal (sin(ωd)) est généré avec la fréquence différentielle ou totale (ωd) résultant du calcul de la différence,
    - dans lequel le signal sinusoïdal (sin(ωd)) est mélangé avec le signal vocal et un signal combiné est généré,
    - dans lequel la distribution de puissance en fonction de la fréquence dans le signal combiné est définie en référence à la fréquence combinée fixe (ωm),
    - dans lequel une valeur de correction (Δω) de la fréquence (ωr) de l'harmonique standard est calculée à partir de la distribution de puissance,
    - dans lequel la fréquence (ωr) de l'harmonique standard est modifiée de la valeur de correction (Δω) et un nouveau mélange est transmis avec la fréquence combinée fixe (ωm) et
    - dans lequel la fréquence fondamentale (ωg) émise est celle qui correspond à la fraction 1/N correspondante de la fréquence (ωr).
  6. Procédé selon la revendication 5, dans lequel, pour déterminer la distribution de puissance, le signal combiné est acheminé à une pluralité de filtres passe-bande (BFn) qui recouvrent les plages de fréquences juxtaposées en étant centrés sur la fréquence combinée fixe.
  7. Procédé selon l'une des revendications 1 à 6, dans lequel l'amplitude de l'au moins un signal de fréquence généré au-dessous de la fréquence limite est déterminée en fonction des amplitudes des signaux de fréquence analysés au-dessus de la fréquence limite.
  8. Procédé selon l'une des revendications 1 à 7, dans lequel la fréquence limite est déterminée en fonction du niveau de bruit.
  9. Procédé selon l'une des revendications 1 à 8, dans lequel le signal vocal est soumis à une libération des signaux parasites avant une conversion dans un spectrogramme.
  10. Application d'un procédé selon l'une des revendications 1 à 9 pour la reproduction d'un signal vocal enregistré dans un véhicule automobile en déplacement.
  11. Application d'un procédé selon l'une des revendications 1 à 9 pour la reproduction d'un signal vocal qui est transmis au moyen d'une liaison téléphonique.
  12. Dispositif pour reconstruire des composantes vocales à basses fréquences à partir de composantes à moyenne fréquence, notamment pour mettre en oeuvre un procédé selon l'une des revendications 1 à 11,
    - comprenant des moyens (4) pour déterminer les composantes de fréquence (ωfa1, ωfa2, ...) au niveau maximum dans le signal vocal au-dessus d'une fréquence limite (ω0) prédéfinie,
    - comprenant des moyens (8) pour mélanger les composantes de fréquence (ωfa1, ωfa2, ...) en vue de déterminer la fréquence fondamentale (ωg) du signal vocal sous la forme de la fréquence différentielle entre à chaque fois deux composantes de fréquence (ωfa1, ωfa2, ...) voisines et
    - comprenant des moyens (12) pour reconstruire le signal vocal au-dessous de la fréquence limite (ω0) à partir de la fréquence fondamentale (ωg) déterminée et du signal vocal.
  13. Dispositif selon la revendication 12, caractérisé en ce que les moyens (12) pour reconstruire le signal vocal au-dessous de la fréquence limite (ω0) déterminent le spectrogramme composé de la fréquence fondamentale (ωg) et des fréquences (ωh1, ωh2, ...) des harmoniques de la fréquence fondamentale (ωg) qui se trouvent au-dessous dé la fréquence limite (ω0) avec une distribution spectrale prédéfinie et une distribution d'amplitude prédéfinie.
  14. Dispositif selon la revendication 12, caractérisé en ce que les moyens (12) présentent un filtre en peigne comprenant une pluralité de filtres passe-bande, les fréquences des filtres passe-bande pouvant être réglées en se basant sur la fréquence fondamentale (ωg) et éventuellement sur une ou plusieurs harmoniques de la fréquence fondamentale (ωg) qui se trouvent au-dessous de la fréquence limite (ω0).
  15. Dispositif selon l'une des revendications 12 à 14, caractérisé en ce que, pour un asservissement adaptatif de la fréquence fondamentale (ωg), sont prévus
    - un élément multiplicateur (24) pour générer la Nième harmonique de la fréquence fondamentale sous la forme de la fréquence (ωr) d'une harmonique standard,
    - un élément mélangeur (28) pour mélanger la fréquence (ωr) de l'harmonique standard avec une fréquence combinée fixe (ωm),
    - un générateur sinusoïdal (30) pour mélanger la fréquence différentielle ou totale (ωd) résultant du mélange,
    - un élément mélangeur (32) pour mélanger le signal sinusoïdal (sin(ωd)) avec lé signal vocal et pour générer un signal combiné,
    - un élément de réglage (34) pour définir la distribution de puissance en fonction de la fréquence dans le signal combiné en référence à la fréquence combinée fixe (ωm) et pour calculer: une valeur de correction (Δω) de la fréquence (ωr) de l'harmonique standard partir de la distribution de puissance,
    - un élément mélangeur (36) pour modifier la fréquence (ωr) de l'harmonique standard de la valeur de correction (Δω) et
    - comprenant un élément multiplicateur (38) pour calculer en tant que fréquence fondamentale (ωg) la fraction 1/N de la fréquence (ωr).
  16. Dispositif selon la revendication 15, caractérisé en ce que l'élément de réglage (34) présente une pluralité de filtres passe-bande qui recouvrent les plages de fréquences juxtaposées en étant centrés sur la fréquence combinée (ωm).
EP01102129A 2000-03-02 2001-02-01 Procédé de reconstruction des basses fréquences du signal de parole à partir de fréquences moyennes Expired - Lifetime EP1130577B1 (fr)

Applications Claiming Priority (2)

Application Number Priority Date Filing Date Title
DE10010037A DE10010037B4 (de) 2000-03-02 2000-03-02 Verfahren zur Rekonstruktion tieffrequenter Sprachanteile aus mittelhohen Frequenzanteilen
DE10010037 2000-03-02

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EP1130577A2 EP1130577A2 (fr) 2001-09-05
EP1130577A3 EP1130577A3 (fr) 2002-09-18
EP1130577B1 true EP1130577B1 (fr) 2007-06-06

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AT (1) ATE364221T1 (fr)
DE (2) DE10010037B4 (fr)

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DE102024101130B3 (de) 2024-01-16 2025-03-20 Cariad Se Verfahren zum Abspielen von digitalen Audiodaten in einem Kraftfahrzeug sowie Kraftfahrzeug

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ATE528748T1 (de) 2006-01-31 2011-10-15 Nuance Communications Inc Verfahren und entsprechendes system zur erweiterung der spektralen bandbreite eines sprachsignals
CN111863006B (zh) * 2019-04-30 2024-07-16 华为技术有限公司 一种音频信号处理方法、音频信号处理装置和耳机
CN112151065B (zh) * 2019-06-28 2024-03-15 力同科技股份有限公司 单音信号频率检测方法、装置、设备及计算机存储介质
CN113362840B (zh) * 2021-06-02 2022-03-29 浙江大学 基于内建传感器欠采样数据的通用语音信息恢复装置及方法

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US4091237A (en) * 1975-10-06 1978-05-23 Lockheed Missiles & Space Company, Inc. Bi-Phase harmonic histogram pitch extractor
US4490843A (en) * 1982-06-14 1984-12-25 Bose Corporation Dynamic equalizing
US4700390A (en) * 1983-03-17 1987-10-13 Kenji Machida Signal synthesizer
DE3782959T2 (de) * 1986-04-01 1993-06-24 Matsushita Electric Ind Co Ltd Erzeuger von niederfrequenten toenen.
JP2779886B2 (ja) * 1992-10-05 1998-07-23 日本電信電話株式会社 広帯域音声信号復元方法

Cited By (1)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
DE102024101130B3 (de) 2024-01-16 2025-03-20 Cariad Se Verfahren zum Abspielen von digitalen Audiodaten in einem Kraftfahrzeug sowie Kraftfahrzeug

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EP1130577A3 (fr) 2002-09-18
ATE364221T1 (de) 2007-06-15
DE10010037A1 (de) 2001-09-06
EP1130577A2 (fr) 2001-09-05
DE10010037B4 (de) 2009-11-26
DE50112581D1 (de) 2007-07-19

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