EP1405302A1 - Method for masking interference during the transfer of digital audio signals - Google Patents

Method for masking interference during the transfer of digital audio signals

Info

Publication number
EP1405302A1
EP1405302A1 EP02740252A EP02740252A EP1405302A1 EP 1405302 A1 EP1405302 A1 EP 1405302A1 EP 02740252 A EP02740252 A EP 02740252A EP 02740252 A EP02740252 A EP 02740252A EP 1405302 A1 EP1405302 A1 EP 1405302A1
Authority
EP
European Patent Office
Prior art keywords
signal
audio signal
digital
data error
audio
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
EP02740252A
Other languages
German (de)
French (fr)
Other versions
EP1405302B1 (en
Inventor
Claus Kupferschmidt
Gerd Penshorn
Arnd Wendland
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.)
Robert Bosch GmbH
Original Assignee
Robert Bosch GmbH
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
Application filed by Robert Bosch GmbH filed Critical Robert Bosch GmbH
Publication of EP1405302A1 publication Critical patent/EP1405302A1/en
Application granted granted Critical
Publication of EP1405302B1 publication Critical patent/EP1405302B1/en
Anticipated expiration legal-status Critical
Expired - Lifetime legal-status Critical Current

Links

Classifications

    • 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
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/005—Correction of errors induced by the transmission channel, if related to the coding algorithm

Definitions

  • the invention is based on a method for masking interference in a reproduced audio signal derived from a digital signal according to the preamble of the independent claim.
  • disturbances in the digital transmission signal occur as a result of non-ideal transmission channels, in particular due to multi-path reception, reflections, shading and attenuation, which have an effect in the form of bit errors. These can be corrected to a certain extent on the transmitter side by a suitable channel coding or by a suitable decoding on the receiver side. If the data error rate within the digital transmission signal rises above a predetermined value, it is no longer possible to correct the bit errors, so that the data content transmitted with the digital transmission signal, for example in the case of a digitally transmitted radio broadcast signal, an audio signal to be reproduced, becomes clear in the form perceivable disturbances.
  • features of the independent patent claim have the advantage that the listener is provided with a reliable assessment basis for the currently selected playback volume for an audio signal transmitted by means of a digital radio signal. This avoids the risk that the user will disadvantageously increase the volume during a weakening or muting of the audio playback due to a high data error rate of the received digital radio signal. In addition, the unpleasant effect of the bit errors within the received digital broadcast signal in the form of the Gurgein within the reproduced audio signal is reduced.
  • the frequency selective signal weakening and substitution caused by bit errors causes chirping disturbances in the audio signal, so-called birdies, so that the subjective perception of the audio signal improves.
  • a particularly good basis for estimating the actually set volume of the digital radio receiver is given by the fact that the superimposition of the substitute signal completely compensates for the attenuation of the audio signal due to a high data error rate, so that the volume of the total audio signal formed from the superimposition of the attenuated audio signal and the substitute signal is the one audio signal received or reproduced without interference.
  • the substitute signal can advantageously be formed in the form of a noise signal, a sine or knowledge tone or a stored or synthesized speech signal. Particularly in the case of a noise signal as a substitute signal, this can furthermore advantageously be adapted in terms of its frequency response to the psychoacoustic properties of the human ear.
  • the substitute signal can be additively superimposed on the attenuated audio signal either in the time domain or in the frequency domain.
  • the method according to the invention is advantageously characterized in that it is basically equally applicable to all audio formats or all audio signals transmitted in digital form, in particular digital broadcast signals of different standards, such as DAB, DSR or the like.
  • the method can be implemented in a particularly simple manner, since the control of both the degree of attenuation of the reproduced audio signal and the degree of superimposition of the substitute signal can be controlled in direct dependence on a data error rate of the received digital radio signal that can be detected by means of data error statistics.
  • the method according to the invention has no effect whatsoever on the source decoding of the audio data from the received digital radio signals, so that the method can also be switched off without influencing the decoded audio signal.
  • FIG. 1 shows a block diagram of an arrangement 1 for carrying out the method according to the invention using the example of an MPEG audio decoder with integrated so-called error concealment, in which an alternative signal is superimposed in the frequency domain on the attenuated audio signal.
  • Figure 2 shows the superposition of audio signal and substitute signal in the frequency domain.
  • MPEG refers to a method developed by the Fraunhofer-Gesellschaft for encoding and compressing digital audio data.
  • the aforementioned audio decoder thus serves to decode the digital audio data in MPEG format.
  • the MPEG-coded digital audio signal 101 which is present at a data input 10 of the arrangement, is fed to a decoder 11.
  • the decoder 11 encodes the encoded digital audio signal and detects and, if necessary, corrects the received data signal.
  • the audio signal 111 present at a first output of the decoder 11 is a filter circuit 12 which, for example in the form of an equalizer, but optionally also in the form of a bandpass filter with adjustable cut-off frequencies, edge steepness and Total gain factor can be formed, supplied.
  • the audio signal 121 evaluated by means of the filter 12 is fed to a superimposition circuit 13 in the present case in the form of an adder 13.
  • the total audio signal 131 which can be taken off at the output of the adder 13 is transformed back in an inverse filter 14 from the frequency range into the time range, so that the total audio signal 141 which can be reproduced via the loudspeakers of an audio system containing the circuit arrangement 1 is present at the output 15 of the circuit arrangement 1.
  • an error signal 112 representing the data error rate of the received digital signal can be taken, which is fed to a circuit arrangement 16 for generating an error statistic.
  • Fault statistics generator 16 can be used to remove an error statistics signal 161 which indicates the data error rate of the digital signal present at the input 10 of the circuit arrangement 1. This is fed to an assignment circuit 17, in which parameters for controlling the equalizer 12 or the filter 12 are selected as a function of the error statistics signal 161. For example, in the case of an approximately undisturbed signal at the data input 10, the equalizer 12 or the filter 12 is controlled via a filter control signal 171 such that the decoded audio signal 111 fed to it can be removed essentially unchanged at the output of the equalizer or filter 12.
  • the assignment circuit 17 is also supplied with a bit error signal 162, likewise generated by the error statistics generator 16, which represents the bit errors of the digital input signal.
  • the bit error signal 162 is derived from the internal checks for frame headers or the data errors themselves and is a direct measure of the current error rate.
  • the error statistics signal 161 is a signal that reacts comparatively slowly to errors in the digital signal.
  • a data record 171 selected as a function of the data error rate or the error statistics signal 161 representing the data error rate, in accordance with a preferred embodiment additionally of the bit error signal 162, for controlling the equalizer 12 or the filter 12 is supplied to the latter by the assignment circuit 17. Furthermore, a data set 172 of filter parameters that is inverse to the selected data set 171 is supplied to an equivalent signal generator 18, which, according to the preferred embodiment of the invention mentioned, is also supplied with the bit error signal 162 from the error statistics generator 16.
  • the substitute signal generator 18 generates depending on the second equalizer or filter parameters 172 supplied to it, in accordance with the preferred embodiment of the invention also in an additional dependency of the bit error signal 162, an equivalent signal shaped according to these parameters, which is fed to a second input of the superimposition circuit 13.
  • a total audio signal 131 can be taken, which results from a superimposition, in the present case an addition, which, according to the first equalizer or Filter parameters 171 by means of the equalizer or filter 12 attenuated audio signal and a replacement signal 181 formed in accordance with the second equalizer or filter parameters 172.
  • the filter parameter set 172 supplied to the substitute signal generation 18 is designed in such a way that the filter curves of the filter 12 and the second filter provided for evaluating the substitute signal in the substitute signal generation 18 compensate each other, so that the result is a linear frequency response .
  • This course of the filter curves can also be seen, for example, in FIG. 2, where the amplitude frequency response 125 of the filter 12 and the further amplitude frequency response 185 of the second filter provided for evaluating the substitute signal in the substitute signal generation 18 are plotted against the frequency 200.
  • the amplitude frequency response 125 of the filter or equalizer 12 which is assigned to a specific degree of error or a specific data error rate of the input signal, decreases from a maximum value with a 3dB cut-off frequency 210 to the value 0.
  • the further frequency response 185 assigned to the same data error rate or data error statistics increases from the value 0 via the 3dB limit frequency 210 to a value which corresponds to the maximum amplitude of the amplitude frequency response 125. Since above a maximum frequency 220, audio signal reproduction for the human ear anyway the further is not perceptible
  • Amplitude frequency response 185 to this maximum frequency 220 down to the value 0.
  • the two frequency responses 125 and 185 of the filter 12 or the substitute signal generator 18 overlap to form an overall linear and constant frequency response.
  • the equivalent signal generator 18 is designed according to a preferred embodiment of the invention in such a way that a neutral noise signal is generated as an equivalent signal.
  • the proportion of the audio signal 121 will increase at the expense of the noise signal 181; in contrast, in the case of an increasing data error rate, the audio signal 121 is increasingly replaced by the noise signal 181.
  • the substitute signal is designed in the form of one or a superposition of several sine or familiar tones.
  • the replacement signal is a stored or synthesized speech signal.
  • the substitute signal 181 can also be designed in the form of a noise that is adapted to the physiology of human hearing and is filtered accordingly.
  • the present method is basically based on any type of digitally coded audio signals applicable. It is within the scope of the present invention that any digitally coded audio signal 101 can be fed to the data input 10. The decoder is then adapted or adapted to the respective type of digitally coded audio signal 101, so that a correctly decoded audio signal 111 can be removed from its output.
  • the invention can also be applied to audio signals present in the time domain, in which case the inverse transformation 14 can be omitted, furthermore filters 12, decoding 16, assignment circuit 17 and substitute signal generation 18 are adapted accordingly.

Landscapes

  • Engineering & Computer Science (AREA)
  • Physics & Mathematics (AREA)
  • Signal Processing (AREA)
  • Health & Medical Sciences (AREA)
  • Audiology, Speech & Language Pathology (AREA)
  • Human Computer Interaction (AREA)
  • Computational Linguistics (AREA)
  • Acoustics & Sound (AREA)
  • Multimedia (AREA)
  • Noise Elimination (AREA)
  • Tone Control, Compression And Expansion, Limiting Amplitude (AREA)
  • Circuit For Audible Band Transducer (AREA)
  • Compression, Expansion, Code Conversion, And Decoders (AREA)
  • Detection And Prevention Of Errors In Transmission (AREA)

Abstract

Disclosed is a method for masking interference in a reproduced audio signal which is derived from a digital signal, wherein the reproduced audio signal is attenuated according to the data error statistics of the digital signal, characterized in that a replacement signal is superimposed on the attenuated audio signal according to the data error statistics of the digital signal. The inventive method ensures in an advantageous manner that a signal can be acoustically reproduced, even in the case of digital input signals that are subjected to large amounts of interference, whereby the adjusted volume in a correspondingly equipped radio receiver can be realistically estimated for a user at any moment, thereby preventing the user from misjudging the real adjusted reproduction volume which, according to prior art, can otherwise occur when the reception signal is subjected to a large amount of interference as a result of an interruption in the audio reproduction.

Description

Verfahren zur Störverdeckung bei digitaler AudiosignalübertragungInterference masking method for digital audio signal transmission
Stand der TechnikState of the art
Die Erfindung geht von einem Verfahren zur Verdeckung von Störungen in einem aus einem digitalen Signal abgeleiteten, wiedergegebenen Audiosignal nach der Gattung des unabhängigen Patentanspruchs aus .The invention is based on a method for masking interference in a reproduced audio signal derived from a digital signal according to the preamble of the independent claim.
In Systemen der digitalen Übertragungstechnik in der mobilen Kommuni ation kommt es infolge nicht idealer Übertragungskanäle, insbesondere durch Mehrwegeempfang, Reflexionen, Abschattungen sowie Dämpfungen, zu Störungen im digitalen Übertragungssignal, die sich in Form von Bitfehlern auswirken. Diese können senderseitig durch eine geeignete Kanalcodierung oder durch eine empfängerseitige geeignete Decodierung in gewissem Umfang korrigiert werden. Steigt die Datenfehlerrate innerhalb des digitalen Übertragungssignals über einen vorgegebenen Wert an, ist eine Korrektur der Bitfehler nicht mehr möglich, so dass sich diese auf die mit dem digitalen Übertragungssignal übertragenen Dateninhalte, beispielsweise im Falle eines digital übertragenen Hörrundfunksignals auf ein wiederzugebendes Audiosignal, in Form deutlich wahrnehmbarer Störungen auswirken. Im Falle analoger Systeme kommt es bei abnehmender Empfangsqualitat zu einer graduellen Verschlechterung der Qualität des im Rundfunksignal enthaltenen Audiosignals, dem beispielsweise analoge FM-Rundfunkempfan er mit einer Stereo-/Mono-Umschaltung bzw. Stummschaltung des wiederzugebenden Audiosignals begegnen.In systems of digital transmission technology in mobile communication, disturbances in the digital transmission signal occur as a result of non-ideal transmission channels, in particular due to multi-path reception, reflections, shading and attenuation, which have an effect in the form of bit errors. These can be corrected to a certain extent on the transmitter side by a suitable channel coding or by a suitable decoding on the receiver side. If the data error rate within the digital transmission signal rises above a predetermined value, it is no longer possible to correct the bit errors, so that the data content transmitted with the digital transmission signal, for example in the case of a digitally transmitted radio broadcast signal, an audio signal to be reproduced, becomes clear in the form perceivable disturbances. In the case of analog systems, there is a gradual deterioration in the quality of the audio signal contained in the radio signal when the quality of reception is reduced, for example the analog FM radio receiver, with a stereo / mono switchover or muting of the audio signal to be reproduced.
Bei digitalen Systemen gibt es eine derartige graduelle oder schleichende Verschlechterung des Signals in Abhängigkeit der Störung des Ubertragungssignals nicht. Vielmehr bewegt sich die Qualität digital übertragener Audiosignale im Bereich entweder einer sehr guten oder einer sehr schlechten Qualität. Um einen gleitenden Übergang von guter zu schlechter Qualität im Falle digitaler Systeme zu realisieren, bedient man sich bei diesen einer Nachbildung dieses Verfahrens (Graceful Degradation) . Die zur Verschleierung von Fehlern dort zum Einsatz kommenden Verfahren neigen wiederum im Falle einer hohen Datenfehlerrate dazu, das Signal leiser bzw. vollkommen stumm zu stellen. Dies kann bei dauerhafter Stummschaltung infolge dauerhaft hoher Datenfehlerrate desIn digital systems, there is no such gradual or gradual deterioration in the signal as a function of the interference in the transmission signal. Rather, the quality of digitally transmitted audio signals is in the range of either a very good or a very poor quality. In order to realize a smooth transition from good to bad quality in the case of digital systems, one uses a replica of this process (graceful degradation). The methods used to conceal errors in turn tend to make the signal quieter or completely mute in the case of a high data error rate. With permanent mute, this can result from a permanently high data error rate
Ubertragungssignals zur Verwirrung des Benutzers fuhren, dem suggeriert wird, dass der Rundfunkempfänger nur ein sehr leises oder überhaupt kein Audiosignal wiedergibt. Dies kann den Benutzer dazu veranlassen, den Lautstarkepegel zur Wiedergabe des Audiosignals über den Lautstarkesteller zu erhohen. Darüber hinaus wird das durch Bitfehler hervorgerufene sogenannte Gurgeln innerhalb des wiedergegebenen Audiosignals in der Regel als sehr unangenehm empfunden. Wird nun das digitale Rundfunksignal nach durch den Benutzer initiierter Lautstarkeerhohung wieder mit ausreichender Qualität, also mit einer Datenfehlerrate, die eine Korrektur der Datenfehler ermöglicht, empfangen, wird die infolge der Empfangsverschlechterung zuvor abgeschwächte oder stummgeschaltete Audiowiedergabe schlagartig wieder aufgenommen, was nach Erhöhung der Wiedergabelautstärke zu einer Schädigung der angeschlossenen Lautsprecher und möglicherweise auch des Gehörs des Benutzers führen kann.Transmission signal lead to confusion of the user, who is suggested that the radio receiver reproduces only a very quiet or no audio signal. This can cause the user to increase the volume level for playing the audio signal through the volume control. In addition, the so-called gurgling caused by bit errors within the reproduced audio signal is generally perceived as very unpleasant. If the digital radio signal is now received again with sufficient quality after the volume increase initiated by the user, that is to say with a data error rate which enables the data errors to be corrected, the result of the Reception deterioration of previously weakened or muted audio playback suddenly resumed, which, after increasing the playback volume, can damage the connected loudspeakers and possibly also the hearing of the user.
Vorteile der ErfindungAdvantages of the invention
Das erfindungs emäße Verfahren mit. en Merkmalen des unabhängigen Patentanspruchs hat demgegenüber den Vorteil, dass dem Zuhörer eine zuverlässige Beurteilungsgrundlage für die momentan vorgewählte Wiedergabelautstärke für ein mittels eines digitalen Rundfunksignals übertragenes Audiosignal vermittelt wird. Damit wird die Gefahr vermieden, dass während einer Abschwächung bzw. Stummschaltung der Audiowiedergabe infolge hoher Datenfehlerrate des empfangenen digitalen Rundfunksignals der Benutzer unvorteilhafterweise die Lautstärke erhöht. Zusätzlich wird die als unangenehm empfundene Wirkung der Bitfehler innerhalb des empfangenen digitalen Rundfunksignals in Form des Gurgeins innerhalb des wiedergegebenen Audiosignals reduziert.The method according to the Invention. In contrast, features of the independent patent claim have the advantage that the listener is provided with a reliable assessment basis for the currently selected playback volume for an audio signal transmitted by means of a digital radio signal. This avoids the risk that the user will disadvantageously increase the volume during a weakening or muting of the audio playback due to a high data error rate of the received digital radio signal. In addition, the unpleasant effect of the bit errors within the received digital broadcast signal in the form of the Gurgein within the reproduced audio signal is reduced.
Dazu wird erfindungsgemäß vorgeschlagen, dass bei einem Verfahren zur Verdeckung von Störungen in einem wiedergegebenen Audiosignal, das aus einem digitalen Signal abgeleitet wird, wobei das wiedergegebene Audiosignal in Abhängigkeit einer Datenfehlerstatistik des digitalen Signals abgeschwächt wird, dem abgeschwächten Audiosignal in Abhängigkeit der Datenfehlerstatistik des digitalen Signals ein Ersatzsignal überlagert wird.For this purpose, it is proposed according to the invention that in a method for masking interference in a reproduced audio signal that is derived from a digital signal, the reproduced audio signal being attenuated as a function of a data error statistic of the digital signal, the attenuated audio signal as a function of the data error statistic of the digital signal a substitute signal is superimposed.
Vorteilhafte Weiterbildungen und Ausführungsformen der Erfindung sind in den abhängigen Patentansprüchen angegeben. So ist es von besonderem Vorteil, dass das wiedergegebene Audiosignal in Abhängigkeit der Datenfehlerstatistik des digitalen Signals frequenzselektiv abgeschwächt wird, und dass das Ersatzsignal frequenzselektiv überlagert wird. Auf diese Weise ist eine weitere Annäherung des Verhaltens eines digitalen Rundfunkempfängers an das eines analogen, insbesondere FM-Rundfunkempfängers angleichbar. So führen analoge FM-Rundfunkempfänger infolge einer Verschlechterung der Empfangsqualität eines empfangenen analogen Rundfunksignals in der Regel einen sogenannten High-Cut, d. h. eine Absenkung hochfrequenter Anteile des wiederzugebenden Audiosignals durch.Advantageous further developments and embodiments of the invention are specified in the dependent claims. So it is particularly advantageous that the reproduced Audio signal depending on the data error statistics of the digital signal is attenuated frequency-selectively, and that the replacement signal is superimposed frequency-selectively. In this way, a further approximation of the behavior of a digital radio receiver to that of an analog, in particular FM radio receiver can be adjusted. Analog FM radio receivers generally carry out a so-called high cut, ie a reduction in high-frequency components of the audio signal to be reproduced, as a result of a deterioration in the reception quality of a received analog radio signal.
Aufgrund der Tatsache, daß Audiosignale im Bereich niedriger Frequenzen eher von tonalen Komponenten geprägt sind und höhere Frequenzbereiche sich eher durch rauschartige Signalanteile auszeichnen, führt die Substitution höherer Frequenzbereiche durch Ersatzrauschen zu einer besseren Signalqualität und damit besserem Hörempfinden nach dem Error-Concealment .Due to the fact that audio signals in the low frequency range are more characterized by tonal components and higher frequency ranges are characterized by noise-like signal components, the substitution of higher frequency ranges with substitute noise leads to better signal quality and thus better hearing perception after error concealment.
Auch werden durch die frequenzselektive Signalabschwächung und -Substitution durch Bitfehler verursachte zwitschernde Störungen im Audiosignal, sogenannte Birdies, reduziert, so daß sich die subjektive Wahrnehmung des Audiosignals verbessert.The frequency selective signal weakening and substitution caused by bit errors causes chirping disturbances in the audio signal, so-called birdies, so that the subjective perception of the audio signal improves.
Eine besonders gute Abschätzungsgrundlage für die tatsächlich eingestellte Lautstärke des digitalen Rundfunkempfängers wird dadurch gegeben, dass die Überlagerung des Ersatzsignals die Abschwächung des Audiosignals infolge einer hohen Datenfehlerrate vollständig kompensiert, so dass die Lautstärke des aus der Überlagerung des abgeschwächten Audiosignals und des Ersatzsignals gebildeten Gesamtaudiosignals der eines ungestört empfangenen bzw. wiedergegebenen Audiosignals entspricht. Das Ersatzsignal kann vorteilhafterweise in Form eines Rauschsignals, eines Sinus- oder Kenntons oder eines gespeicherten oder synthetisierten Sprachsignals gebildet sein. Insbesondere im Falle eines Rauschsignals als Ersatzsignal kann dieses weiterhin vorteilhafterweise hinsichtlich seines Frequenzgangs an die psychoakustischen Eigenschaften des menschlichen Gehörs angepaßt sein.A particularly good basis for estimating the actually set volume of the digital radio receiver is given by the fact that the superimposition of the substitute signal completely compensates for the attenuation of the audio signal due to a high data error rate, so that the volume of the total audio signal formed from the superimposition of the attenuated audio signal and the substitute signal is the one audio signal received or reproduced without interference. The substitute signal can advantageously be formed in the form of a noise signal, a sine or knowledge tone or a stored or synthesized speech signal. Particularly in the case of a noise signal as a substitute signal, this can furthermore advantageously be adapted in terms of its frequency response to the psychoacoustic properties of the human ear.
Weiterhin kann das Ersatzsignal dem abgeschwächten Audiosignal entweder im Zeitbereich oder im Frequenzbereich additiv überlagert werden.Furthermore, the substitute signal can be additively superimposed on the attenuated audio signal either in the time domain or in the frequency domain.
Das erfindungsgemäße Verfahren zeichnet sich in vorteilhafter Weise dadurch aus, dass es grundsätzlich auf alle Audioformate bzw. alle in digitaler Form übertragene Audiosignale, insbesondere digitale Rundfunksignale verschiedener Standards, wie beispielsweise DAB, DSR oder ähnliche, gleichermaßen anwendbar ist.The method according to the invention is advantageously characterized in that it is basically equally applicable to all audio formats or all audio signals transmitted in digital form, in particular digital broadcast signals of different standards, such as DAB, DSR or the like.
Darüber hinaus ist das Verfahren besonders einfach realisierbar, da die Steuerung sowohl des Maßes der Abschwächung des wiedergegebenen Audiosignals, als auch des Maßes der Überlagerung des Ersatzsignals in direkter Abhängigkeit einer mittels einer Datenfehlerstatistik erfaßbaren Datenfehlerrate des empfangenen digitalen Rundfunksignals steuerbar ist.In addition, the method can be implemented in a particularly simple manner, since the control of both the degree of attenuation of the reproduced audio signal and the degree of superimposition of the substitute signal can be controlled in direct dependence on a data error rate of the received digital radio signal that can be detected by means of data error statistics.
Darüber hinaus ist es von besonderem Vorteil, dass das erfindungsgemäße Verfahren sich in keiner Weise auf die Quellendecodierung der Audiodaten aus den empfangenen digitalen Rundfunksignalen auswirkt, so dass das Verfahren ohne Beeinflussung des decodierten Audiosignals auch abschaltbar ist. ZeichnungenIn addition, it is particularly advantageous that the method according to the invention has no effect whatsoever on the source decoding of the audio data from the received digital radio signals, so that the method can also be switched off without influencing the decoded audio signal. drawings
Ein vorteilhaftes Ausführungsbeispiel der Erfindung ist in den Figuren dargestellt und nachfolgend näher erläutert.An advantageous embodiment of the invention is shown in the figures and explained in more detail below.
Figur 1 zeigt ein Blockschaltbild einer Anordnung 1 zur Durchführung des erfindungsgemäßen Verfahrens am Beispiel eines MPEG-Audiodecoders mit integriertem sogenanntem Error- Concealment, bei dem ein Ersatzsignal dem bedarfsweise abgeschwächten Audiosignal im Frequenzbereich überlagert wird.FIG. 1 shows a block diagram of an arrangement 1 for carrying out the method according to the invention using the example of an MPEG audio decoder with integrated so-called error concealment, in which an alternative signal is superimposed in the frequency domain on the attenuated audio signal.
Figur 2 zeigt die Überlagerung von Audiosignal und Ersatzsignal im Frequenzbereich.Figure 2 shows the superposition of audio signal and substitute signal in the frequency domain.
Beschreibung der AusführungsbeispieleDescription of the embodiments
In Figur 1 ist ein Audiodecoder MPEG 1, 2 Layer 2 mit integrierter Bit- bzw. Datenfehlerverschleierung dargestellt. MPEG bezeichnet dabei ein von der Fraunhofer- Gesellschaft entwickeltes Verfahren zur Codierung bzw. Komprimierung digitaler Audiodaten. Der genannte Audiodecoder dient somit der Decodierung der im MPEG-Format vorliegenden digitalen Audiodaten.1 shows an audio decoder MPEG 1, 2 Layer 2 with integrated bit or data error concealment. MPEG refers to a method developed by the Fraunhofer-Gesellschaft for encoding and compressing digital audio data. The aforementioned audio decoder thus serves to decode the digital audio data in MPEG format.
Das MPEG-codierte digitale Audiosignal 101, das an einem Dateneingang 10 der Anordnung ansteht, ist einem Decoder 11 zugeführt. Im Decoder 11 erfolgt die Decodierung des codierten digitalen Audiosignals sowie eine Fehlererkennung und gegebenenfalls -korrektur des empfangenen Datensignals. Das an einem ersten Ausgang des Decoders 11 anstehende Audiosignal 111 ist einer Filterschaltung 12, die beispielsweise in Form eines Equalizers, wahlweise aber auch in Form eines Bandpassfilters mit einstellbaren Grenzfrequenzen, Flankensteilheit und Gesamtverstärkungsfaktor ausgebildet sein kann, zugeführt. Das mittels des Filters 12 bewertete Audiosignal 121 ist einer Überlagerungsschaltung 13 im vorliegenden Fall in Form eines Addiergliedes 13 zugeführt. Das am Ausgang des Addiergliedes 13 abnehmbare Gesamtaudiosignal 131 wird in einem inversen Filter 14 vom Frequenz- in den Zeitbereich rücktransformiert, so dass am Ausgang 15 der Schaltungsanordnung 1 das über die Lautsprecher einer die Schaltungsanordnung 1 enthaltenden Audioanlage wiedergebbare Gesamtaudiosignal 141 ansteht.The MPEG-coded digital audio signal 101, which is present at a data input 10 of the arrangement, is fed to a decoder 11. The decoder 11 encodes the encoded digital audio signal and detects and, if necessary, corrects the received data signal. The audio signal 111 present at a first output of the decoder 11 is a filter circuit 12 which, for example in the form of an equalizer, but optionally also in the form of a bandpass filter with adjustable cut-off frequencies, edge steepness and Total gain factor can be formed, supplied. The audio signal 121 evaluated by means of the filter 12 is fed to a superimposition circuit 13 in the present case in the form of an adder 13. The total audio signal 131 which can be taken off at the output of the adder 13 is transformed back in an inverse filter 14 from the frequency range into the time range, so that the total audio signal 141 which can be reproduced via the loudspeakers of an audio system containing the circuit arrangement 1 is present at the output 15 of the circuit arrangement 1.
Die Notwendigkeit einer Rücktransformation 14 ergibt sich aus der Tatsache, daß MPEG-codierte Signale im Frequenzbereich vorliegen, jeder Abtastwert des Audiosignals liegt somit in Form dessen Spektralverteilung vor.The need for a reverse transformation 14 arises from the fact that MPEG-coded signals are present in the frequency domain, and each sample of the audio signal is therefore in the form of its spectral distribution.
An einem zweiten Ausgang des Decoders 11 ist ein die Datenfehlerrate des empfangenen digitalen Signals repräsentierendes Fehlersignal 112 abnehmbar, das einer Schaltungsanordnung 16 zur Erzeugung einer Fehlerstatistik zugeführt ist. An einem ersten Ausgang derAt a second output of the decoder 11, an error signal 112 representing the data error rate of the received digital signal can be taken, which is fed to a circuit arrangement 16 for generating an error statistic. At a first exit of the
Fehlerstatistikerzeugung 16 ist ein die Datenfehlerrate des am Eingang 10 der Schaltungsanordnung 1 anstehenden digitalen Signals anzeigendes Fehlerstatistiksignal 161 abnehmbar. Dieses ist einer ZuordnungsSchaltung 17 zugeführt, in der in Abhängigkeit des Fehlerstatistiksignals 161 Parameter zur Steuerung des Equalizers 12 bzw. des Filters 12 ausgewählt werden. Beispielsweise wird im Falle eines näherungsweise ungestörten Signals am Dateneingang 10 der Equalizer 12 bzw. das Filter 12 über ein Filtersteuersignal 171 derart gesteuert, dass das diesem zugeführte decodierte Audiosignal 111 im wesentlichen unverändert am Ausgang des Equalizers bzw. Filters 12 abnehmbar ist. Demgegenüber wird bei zunehmender Datenfehlerrate in der Zuordnungsschaltung 17 ein Parametersatz zur Steuerung des Equalizers 12 bzw. Filters 12 dergestalt ausgewählt, dass zunächst höherfrequente Anteile des Audiosignals 111, mit weiter zunehmender Datenfehlerrate zunehmend auch niederfrequente Anteile des Audiosignals 111 und schließlich das gesamte Audiosignal abgeschwächt wird.Fault statistics generator 16 can be used to remove an error statistics signal 161 which indicates the data error rate of the digital signal present at the input 10 of the circuit arrangement 1. This is fed to an assignment circuit 17, in which parameters for controlling the equalizer 12 or the filter 12 are selected as a function of the error statistics signal 161. For example, in the case of an approximately undisturbed signal at the data input 10, the equalizer 12 or the filter 12 is controlled via a filter control signal 171 such that the decoded audio signal 111 fed to it can be removed essentially unchanged at the output of the equalizer or filter 12. In contrast, as the data error rate increases in the allocation circuit 17 Parameter set for controlling the equalizer 12 or filter 12 selected such that initially higher-frequency components of the audio signal 111, with a further increasing data error rate also increasingly low-frequency components of the audio signal 111 and finally the entire audio signal is attenuated.
Gemäß einer bevorzugten Ausführungsform der Erfindung ist der Zuordnungsschaltung 17 weiterhin ein ebenfalls von der Fehlerstatistikerzeugung 16 generiertes Bitfehlersignal 162 zugeführt, das die Bitfehler des digitalen Eingangssignals repräsentiert. Das Bitfehlersignal 162 wird aus den internen Prüfungen für Rahmenheader oder der Datenfehler selbst abgeleitet und ist ein direktes Maß für die aktuelle Fehlerrate. Demgegenüber ist das Fehlerstatistiksignal 161 aufgrund einer Tiefpaßcharakteristik ein auf Fehler im digitalen Signal vergleichsweise langsam reagierendes Signal .According to a preferred embodiment of the invention, the assignment circuit 17 is also supplied with a bit error signal 162, likewise generated by the error statistics generator 16, which represents the bit errors of the digital input signal. The bit error signal 162 is derived from the internal checks for frame headers or the data errors themselves and is a direct measure of the current error rate. In contrast, due to a low-pass characteristic, the error statistics signal 161 is a signal that reacts comparatively slowly to errors in the digital signal.
Eine in Abhängigkeit der Datenfehlerrate bzw. des die Datenfehlerrate repräsentierenden Fehlerstatistiksignals 161, gemäß einer bevorzugten Ausführungsform zusätzlich des Bitfehlersignals 162, ausgewählter Datensatz 171 zur Steuerung des Equalizers 12 bzw. des Filters 12 ist diesem von der ZuordnungsSchaltung 17 zugeführt. Des weiteren ist ein dem gewählten Datensatz 171 inverser Datensatz 172 an Filterparametern einem Ersatzsignalgenerator 18 zugeführt, dem weiterhin, gemäß erwähnter bevorzugter Ausführungsform der Erfindung, das Bitfehlersignal 162 von der Fehlerstatistikerzeugung 16 zugeführt ist.A data record 171 selected as a function of the data error rate or the error statistics signal 161 representing the data error rate, in accordance with a preferred embodiment additionally of the bit error signal 162, for controlling the equalizer 12 or the filter 12 is supplied to the latter by the assignment circuit 17. Furthermore, a data set 172 of filter parameters that is inverse to the selected data set 171 is supplied to an equivalent signal generator 18, which, according to the preferred embodiment of the invention mentioned, is also supplied with the bit error signal 162 from the error statistics generator 16.
Der Ersatzsignalgenerator 18 erzeugt in Abhängigkeit der ihm zugeführten zweiten Equalizer- bzw. Filterparameter 172, gemäß der bevorzugten Ausführungsform der Erfindung darüber hinaus in zusätzlicher Abhängigkeit des Bitfehlersignals 162, ein entsprechend diesen Parametern geformtes Ersatzsignal, das einem zweiten Eingang der Überlagerungsschaltung 13 zugeführt ist. Somit ist am Ausgang der Überlagerungsschaltung 13 ein Gesamtaudiosignal 131 abnehmbar, das aus einer Überlagerung, im vorliegendem Fall einer Addition, des nach Maßgabe der ersten Equalizerbzw. Filterparameter 171 mittels des Equalizers oder Filters 12 abgeschwächten Audiosignals und eines nach Maßgabe der zweiten Equalizer- bzw. Filterparameter 172 geformten Ersatzsignals 181 besteht.The substitute signal generator 18 generates depending on the second equalizer or filter parameters 172 supplied to it, in accordance with the preferred embodiment of the invention also in an additional dependency of the bit error signal 162, an equivalent signal shaped according to these parameters, which is fed to a second input of the superimposition circuit 13. Thus, at the output of the superimposition circuit 13, a total audio signal 131 can be taken, which results from a superimposition, in the present case an addition, which, according to the first equalizer or Filter parameters 171 by means of the equalizer or filter 12 attenuated audio signal and a replacement signal 181 formed in accordance with the second equalizer or filter parameters 172.
Der der Ersatzsignalgenerierung 18 zugeführte Filterparametersatz 172 ist gemass einer bevorzugten Ausführungsform der Erfindung derart ausgelegt, dass sich die Filterkurven des Filters 12 und des zur Bewertung des Ersatzsignals in der Ersatzsignalgenerierung 18 vorgesehenen zweiten Filters gegenseitig kompensieren, so dass sich in der Summe ein linearer Frequenzgang ergibt. Dieser Verlauf der Filterkurven ist beispielsweise auch Figur 2 zu entnehmen, wo der Amplitudenfrequenzgang 125 des Filters 12 und der weitere Amplitudenfrequenzgang 185 des zur Bewertung des Ersatzsignals in der Ersatzsignalgenerierung 18 vorgesehenen zweiten Filters über der Frequenz 200 aufgetragen sind. Wie der Figur zu entnehmen, nimmt der Amplitudenfrequenzgang 125 des Filters bzw. Equalizers 12, der einem bestimmten Fehlergrad bzw. einer bestimmten Datenfehlerrate des Eingangssignals zugeordnet ist, von einem Maximalwert mit einer 3dB-Grenzfrequenz 210 schließt sich auf den Wert 0 ab. Demgegenüber nimmt der derselben Datenfehlerrate bzw. Datenfehlerstatistik zugeordnete weitere Frequenzgang 185 vom Wert 0 über die 3dB- Grenzfrequenz 210 auf einen Wert zu, der der maximalen Amplitude des Amplitudenfrequenzgangs 125 entspricht. Da oberhalb einer Maximalfrequenz 220 eine Audiosignalwiedergabe ohnehin für das menschliche Gehör nicht wahrnehmbar ist, fällt der weitereAccording to a preferred embodiment of the invention, the filter parameter set 172 supplied to the substitute signal generation 18 is designed in such a way that the filter curves of the filter 12 and the second filter provided for evaluating the substitute signal in the substitute signal generation 18 compensate each other, so that the result is a linear frequency response , This course of the filter curves can also be seen, for example, in FIG. 2, where the amplitude frequency response 125 of the filter 12 and the further amplitude frequency response 185 of the second filter provided for evaluating the substitute signal in the substitute signal generation 18 are plotted against the frequency 200. As can be seen from the figure, the amplitude frequency response 125 of the filter or equalizer 12, which is assigned to a specific degree of error or a specific data error rate of the input signal, decreases from a maximum value with a 3dB cut-off frequency 210 to the value 0. In contrast, the further frequency response 185 assigned to the same data error rate or data error statistics increases from the value 0 via the 3dB limit frequency 210 to a value which corresponds to the maximum amplitude of the amplitude frequency response 125. Since above a maximum frequency 220, audio signal reproduction for the human ear anyway the further is not perceptible
Amplitudenfrequenzgang 185 zu dieser Maximalfrequenz 220 hin auf den Wert 0 ab.Amplitude frequency response 185 to this maximum frequency 220 down to the value 0.
Wie Figur 2 zu entnehmen, überlagern sich die beiden Frequenzgänge 125 und 185 des Filters 12 bzw. der Ersatzsignalerzeugung 18 zu einem insgesamt linearen und konstanten Frequenzgang.As can be seen in FIG. 2, the two frequency responses 125 and 185 of the filter 12 or the substitute signal generator 18 overlap to form an overall linear and constant frequency response.
Der Ersatzsignalgenerator 18 ist ge ass einer bevorzugten Ausführungsform der Erfindung derart ausgelegt, dass in diesem ein neutrales Rauschsignal als Ersatzsignal erzeugt wird. Somit ergibt sich am Ausgang 15 der Schaltung 1 der Figur 1 ein Gesamtaudiosignal 141, das aus einer Überlagerung eines gemass der gemessenen Datenfehlerrate abgeschwächten Audiosignals und eines ebenfalls gemass der Datenfehlerrate erzeugten Rauschsignals besteht. Zu niedrigeren Datenfehlerraten hin wird der Anteil des Audiosignals 121 auf Kosten des Rauschsignals 181 zunehmen, demgegenüber wird im Falle zunehmender Datenfehlerrate das Audiosignal 121 zunehmend durch das Rauschsignal 181 ersetzt .The equivalent signal generator 18 is designed according to a preferred embodiment of the invention in such a way that a neutral noise signal is generated as an equivalent signal. This results in an overall audio signal 141 at the output 15 of the circuit 1 in FIG. 1, which consists of a superimposition of an audio signal attenuated according to the measured data error rate and a noise signal also generated according to the data error rate. Towards lower data error rates, the proportion of the audio signal 121 will increase at the expense of the noise signal 181; in contrast, in the case of an increasing data error rate, the audio signal 121 is increasingly replaced by the noise signal 181.
Ge ass einer vorteilhafter Weiterbildungen der Erfindung kann es demgegenüber vorgesehen sein, dass das Ersatzsignal in Form eines oder einer Überlagerung mehrerer Sinus- oder Kenntöne ausgebildet ist. Des weiteren kann es vorgesehen sein, dass das Ersatzsignal ein gespeichertes oder synthetisiertes Sprachsignal ist. Weiter kann das Ersatzsignal 181 auch in Form eines an die Physiologie des menschlichen Gehörs angepassten und entsprechend gefilterten Rauschens ausgeführt sein.In contrast, according to an advantageous further development of the invention, it can be provided that the substitute signal is designed in the form of one or a superposition of several sine or familiar tones. Furthermore, it can be provided that the replacement signal is a stored or synthesized speech signal. Furthermore, the substitute signal 181 can also be designed in the form of a noise that is adapted to the physiology of human hearing and is filtered accordingly.
Wie eingangs erwähnt, ist das vorliegende Verfahren grundsätzlich auf jedwede Art digital codierter Audiosignale anwendbar. So liegt es im Bereich vorliegender Erfindung, dass dem Dateneingang 10 ein beliebiges digital codiertes Audiosignal 101 zuführbar ist. Der Decodierer ist dann an die jeweilige Art des digital codierten Audiosignals 101 angepasst bzw. anzupassen, so dass an dessen Ausgang ein korrekt decodiertes Audiosignal 111 abnehmbar ist.As mentioned at the beginning, the present method is basically based on any type of digitally coded audio signals applicable. It is within the scope of the present invention that any digitally coded audio signal 101 can be fed to the data input 10. The decoder is then adapted or adapted to the respective type of digitally coded audio signal 101, so that a correctly decoded audio signal 111 can be removed from its output.
Grundsätzlich ist die Erfindung auch auf im Zeitbereich vorliegende Audiosignale anwendbar, für diesen Fall kann dann die Rücktransformation 14 entfallen, weiters sind dann Filter 12, Decodierung 16, Zuordnungsschaltung 17 und Ersatzsignalgenerierung 18 entsprechend angepasst. In principle, the invention can also be applied to audio signals present in the time domain, in which case the inverse transformation 14 can be omitted, furthermore filters 12, decoding 16, assignment circuit 17 and substitute signal generation 18 are adapted accordingly.

Claims

Ansprüche Expectations
1. Verfahren zur Verdeckung von Störungen in einem wiedergegebenen Audiosignal, das aus einem digitalen Signal abgeleitet wird, wobei das wiedergegebene Audiosignal in Abhängigkeit einer1. A method for masking interference in a reproduced audio signal which is derived from a digital signal, the reproduced audio signal depending on a
Datenfehlerstatistik des digitalen Signals abgeschwächt wird, dadurch gekennzeichnet, daß dem abgeschwächten Audiosignal in Abhängigkeit derData error statistics of the digital signal is attenuated, characterized in that the attenuated audio signal depending on the
Datenfehlerstatistik des digitalen Signals ein Ersatzsignal überlagert wird.Data error statistics of the digital signal is superimposed on an equivalent signal.
2. Verfahren nach Anspruch 1, dadurch gekennzeichnet, daß das wiedergegebene Audiosignal in Abhängigkeit der Datenfehlerstatistik des digitalen Signals frequenzselektiv abgeschwächt wird, und daß das Ersatzsignal frequenzselektiv überlagert wird.2. The method according to claim 1, characterized in that the reproduced audio signal is attenuated in a frequency-selective manner as a function of the data error statistics of the digital signal, and in that the replacement signal is frequency-selectively superimposed.
3. Verfahren nach Anspruch 1 oder 2, dadurch gekennzeichnet, daß die Überlagerung des Ersatzsignals die Abschwächung des Audiosignals kompensiert .3. The method according to claim 1 or 2, characterized in that the superposition of the substitute signal compensates for the attenuation of the audio signal.
4. Verfahren nach einem der vorhergehenden Ansprüche, dadurch gekennzeichnet, daß das Ersatzsignal durch ein Rauschsignal, einen Sinus- oder Kennton oder durch ein Sprachsignal gebildet wird. 4. The method according to any one of the preceding claims, characterized in that the replacement signal is formed by a noise signal, a sine or characteristic tone or by a speech signal.
EP02740252A 2001-06-22 2002-04-12 Method for masking interference during the transfer of digital audio signals Expired - Lifetime EP1405302B1 (en)

Applications Claiming Priority (3)

Application Number Priority Date Filing Date Title
DE10130233 2001-06-22
DE10130233A DE10130233A1 (en) 2001-06-22 2001-06-22 Interference masking method for digital audio signal transmission
PCT/DE2002/001368 WO2003001509A1 (en) 2001-06-22 2002-04-12 Method for masking interference during the transfer of digital audio signals

Publications (2)

Publication Number Publication Date
EP1405302A1 true EP1405302A1 (en) 2004-04-07
EP1405302B1 EP1405302B1 (en) 2004-12-15

Family

ID=7689138

Family Applications (1)

Application Number Title Priority Date Filing Date
EP02740252A Expired - Lifetime EP1405302B1 (en) 2001-06-22 2002-04-12 Method for masking interference during the transfer of digital audio signals

Country Status (6)

Country Link
US (1) US20040221209A1 (en)
EP (1) EP1405302B1 (en)
JP (1) JP4221288B2 (en)
DE (2) DE10130233A1 (en)
ES (1) ES2233828T3 (en)
WO (1) WO2003001509A1 (en)

Families Citing this family (8)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
KR100708123B1 (en) * 2005-02-04 2007-04-16 삼성전자주식회사 Method and apparatus for controlling audio volume automatically
WO2007025561A1 (en) * 2005-09-01 2007-03-08 Telefonaktiebolaget Lm Ericsson (Publ) Processing encoded real-time data
US8620644B2 (en) * 2005-10-26 2013-12-31 Qualcomm Incorporated Encoder-assisted frame loss concealment techniques for audio coding
DE102006034625A1 (en) * 2006-07-27 2008-01-31 Bayerische Motoren Werke Ag Sound releasing method for mobile radio device, involves releasing audible spare signal on output device in case of low quality of radio signal received by receiver unit
JP2013031151A (en) 2011-06-20 2013-02-07 Renesas Electronics Corp Encryption communication system and encryption communication method
BR112016027898B1 (en) 2014-06-13 2023-04-11 Telefonaktiebolaget Lm Ericsson (Publ) METHOD, ENTITY OF RECEIPT, AND, NON-TRANSITORY COMPUTER READABLE STORAGE MEDIA FOR HIDING FRAME LOSS
JP2016046719A (en) * 2014-08-25 2016-04-04 株式会社東芝 Data generation device, communication device, mobile body, data generation method, and program
FR3056043B1 (en) * 2016-09-15 2019-02-01 Continental Automotive France DEVICE FOR PROCESSING AN AUDIO SIGNAL FROM A RADIO FREQUENCY SIGNAL

Family Cites Families (9)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
JPS5938912A (en) * 1982-08-27 1984-03-03 Nec Corp Pcm audio error compensating circuit
CA2137459A1 (en) * 1993-05-04 1994-11-10 Stephen V. Cahill Apparatus and method for substantially eliminating noise in an audible output signal
FI98164C (en) * 1994-01-24 1997-04-25 Nokia Mobile Phones Ltd Processing of speech coder parameters in a telecommunication system receiver
DE4427351C1 (en) * 1994-08-02 1996-03-28 Siemens Ag Signal processing method and arrangement for block coded speech signals of a message system
JP3264822B2 (en) * 1995-04-05 2002-03-11 三菱電機株式会社 Mobile communication equipment
FI963870A (en) * 1996-09-27 1998-03-28 Nokia Oy Ab Masking errors in a digital audio receiver
US6032048A (en) * 1997-03-17 2000-02-29 Ericsson Inc. Method and apparatus for compensating for click noise in an FM receiver
JP3649854B2 (en) * 1997-05-09 2005-05-18 松下電器産業株式会社 Speech encoding device
US6915263B1 (en) * 1999-10-20 2005-07-05 Sony Corporation Digital audio decoder having error concealment using a dynamic recovery delay and frame repeating and also having fast audio muting capabilities

Non-Patent Citations (1)

* Cited by examiner, † Cited by third party
Title
See references of WO03001509A1 *

Also Published As

Publication number Publication date
WO2003001509A1 (en) 2003-01-03
JP2004533021A (en) 2004-10-28
US20040221209A1 (en) 2004-11-04
DE50201804D1 (en) 2005-01-20
ES2233828T3 (en) 2005-06-16
JP4221288B2 (en) 2009-02-12
DE10130233A1 (en) 2003-01-02
EP1405302B1 (en) 2004-12-15

Similar Documents

Publication Publication Date Title
DE3639753C2 (en)
DE19921122C1 (en) Method and device for concealing an error in a coded audio signal and method and device for decoding a coded audio signal
US5682461A (en) Method of transmitting or storing digitalized, multi-channel audio signals
DE69325431T2 (en) High-performance coding device and device and method for noise spectrum modification
EP0978172B1 (en) Method for masking defects in a stream of audio data
DE102007029381A1 (en) Digital signal e.g. audio signal, processing device, has decision section, which assumes forecast data before deletion as interpolation data, when absolute value is lower than resolution
EP1405302B1 (en) Method for masking interference during the transfer of digital audio signals
EP0775360B1 (en) Signal-processing method and device for substituting non-correctable blocks in a receiver for block-coded audio signals
DE4343366C2 (en) Method and circuit arrangement for increasing the bandwidth of narrowband speech signals
EP1245024B1 (en) Method for the error concealment of digital audio data by means of spectral equalisation
EP0378609B1 (en) Process for transmitting an audio signal
EP1303856B1 (en) Method for concealing transmission errors in digital audio data
DE2338482B2 (en) Circuit arrangement for noise reduction of a stereo broadcast broadcast according to the FCC system on the receiver side, combined with a mono-stereo display
EP1274175B1 (en) Method of checking the reception of alternative reception frequencies in a car radio and car radio
DE19515805C2 (en) Method for data reduction of a digitized audio signal
DE102008002689A1 (en) Apparatus and method for adjusting the reproduction spectrum of two audio sources
DE19818609C2 (en) Method and device for noise filtering
EP1672823A1 (en) Receiving apparatus for digital radio broadcasting systems
DE3149803A1 (en) Circuit arrangement for automatic dynamic compression
DE19735675A1 (en) Method for concealing errors in an audio data stream
DE19734529A1 (en) Replay volume control method for radio receiver
DE10113322A1 (en) Encoding audio data involves comparing required bit number per frame with desired bit number per frame, adapting quality measure for quantisation depending on comparison result
DE102005035094A1 (en) Audio signal reproducing device

Legal Events

Date Code Title Description
PUAI Public reference made under article 153(3) epc to a published international application that has entered the european phase

Free format text: ORIGINAL CODE: 0009012

17P Request for examination filed

Effective date: 20040122

AK Designated contracting states

Kind code of ref document: A1

Designated state(s): AT BE CH CY DE DK ES FI FR GB GR IE IT LI LU MC NL PT SE TR

GRAP Despatch of communication of intention to grant a patent

Free format text: ORIGINAL CODE: EPIDOSNIGR1

GRAS Grant fee paid

Free format text: ORIGINAL CODE: EPIDOSNIGR3

GRAA (expected) grant

Free format text: ORIGINAL CODE: 0009210

AK Designated contracting states

Kind code of ref document: B1

Designated state(s): DE ES FR GB IT

REG Reference to a national code

Ref country code: GB

Ref legal event code: FG4D

Free format text: NOT ENGLISH

REG Reference to a national code

Ref country code: IE

Ref legal event code: FG4D

Free format text: GERMAN

REF Corresponds to:

Ref document number: 50201804

Country of ref document: DE

Date of ref document: 20050120

Kind code of ref document: P

GBT Gb: translation of ep patent filed (gb section 77(6)(a)/1977)

Effective date: 20050406

REG Reference to a national code

Ref country code: ES

Ref legal event code: FG2A

Ref document number: 2233828

Country of ref document: ES

Kind code of ref document: T3

PLBE No opposition filed within time limit

Free format text: ORIGINAL CODE: 0009261

STAA Information on the status of an ep patent application or granted ep patent

Free format text: STATUS: NO OPPOSITION FILED WITHIN TIME LIMIT

26N No opposition filed

Effective date: 20050916

ET Fr: translation filed
PGFP Annual fee paid to national office [announced via postgrant information from national office to epo]

Ref country code: ES

Payment date: 20120423

Year of fee payment: 11

PGFP Annual fee paid to national office [announced via postgrant information from national office to epo]

Ref country code: GB

Payment date: 20130422

Year of fee payment: 12

PGFP Annual fee paid to national office [announced via postgrant information from national office to epo]

Ref country code: IT

Payment date: 20130424

Year of fee payment: 12

GBPC Gb: european patent ceased through non-payment of renewal fee

Effective date: 20140412

PG25 Lapsed in a contracting state [announced via postgrant information from national office to epo]

Ref country code: GB

Free format text: LAPSE BECAUSE OF NON-PAYMENT OF DUE FEES

Effective date: 20140412

PG25 Lapsed in a contracting state [announced via postgrant information from national office to epo]

Ref country code: IT

Free format text: LAPSE BECAUSE OF NON-PAYMENT OF DUE FEES

Effective date: 20140412

REG Reference to a national code

Ref country code: ES

Ref legal event code: FD2A

Effective date: 20150527

PG25 Lapsed in a contracting state [announced via postgrant information from national office to epo]

Ref country code: ES

Free format text: LAPSE BECAUSE OF NON-PAYMENT OF DUE FEES

Effective date: 20140413

REG Reference to a national code

Ref country code: FR

Ref legal event code: PLFP

Year of fee payment: 15

REG Reference to a national code

Ref country code: DE

Ref legal event code: R084

Ref document number: 50201804

Country of ref document: DE

REG Reference to a national code

Ref country code: FR

Ref legal event code: PLFP

Year of fee payment: 16

PGFP Annual fee paid to national office [announced via postgrant information from national office to epo]

Ref country code: FR

Payment date: 20170424

Year of fee payment: 16

PG25 Lapsed in a contracting state [announced via postgrant information from national office to epo]

Ref country code: FR

Free format text: LAPSE BECAUSE OF NON-PAYMENT OF DUE FEES

Effective date: 20180430

PGFP Annual fee paid to national office [announced via postgrant information from national office to epo]

Ref country code: DE

Payment date: 20190627

Year of fee payment: 18

REG Reference to a national code

Ref country code: DE

Ref legal event code: R119

Ref document number: 50201804

Country of ref document: DE

PG25 Lapsed in a contracting state [announced via postgrant information from national office to epo]

Ref country code: DE

Free format text: LAPSE BECAUSE OF NON-PAYMENT OF DUE FEES

Effective date: 20201103