EP1258865B1 - Device for improving the intelligibility of audio signals containing speech - Google Patents

Device for improving the intelligibility of audio signals containing speech Download PDF

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Publication number
EP1258865B1
EP1258865B1 EP02005495A EP02005495A EP1258865B1 EP 1258865 B1 EP1258865 B1 EP 1258865B1 EP 02005495 A EP02005495 A EP 02005495A EP 02005495 A EP02005495 A EP 02005495A EP 1258865 B1 EP1258865 B1 EP 1258865B1
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EP
European Patent Office
Prior art keywords
circuit arrangement
input
signal
pass filter
arrangement according
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EP02005495A
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German (de)
French (fr)
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EP1258865A3 (en
EP1258865A2 (en
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Matthias Vierthaler
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TDK Micronas GmbH
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TDK Micronas GmbH
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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
    • G10L21/0316—Speech enhancement, e.g. noise reduction or echo cancellation by changing the amplitude
    • G10L21/0364—Speech enhancement, e.g. noise reduction or echo cancellation by changing the amplitude for improving intelligibility
    • 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
    • G10L21/0208—Noise filtering
    • G10L21/0216—Noise filtering characterised by the method used for estimating noise
    • G10L21/0232—Processing in the frequency domain
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R2225/00—Details of deaf aids covered by H04R25/00, not provided for in any of its subgroups
    • H04R2225/43—Signal processing in hearing aids to enhance the speech intelligibility

Definitions

  • the invention relates to a circuit arrangement for improving the intelligibility of speech-containing audio signals according to the features of the preamble of claim 1.
  • the aim of the present invention is to improve the speech intelligibility of a relatively good audio signal with unchanged volume. This means equal intelligibility at lower volume or improved intelligibility in ambient noise.
  • consonants are about 12 dB weaker than vowels. Increasing the consonants relative to the vowels increases the intelligibility of speech in the audio signal. If you replace the clipper with a fast "peak limiter” (22 msec) you can increase the intelligibility even further. At -10 dBlimitting, intelligibility increased from 56% to 84%.
  • US 5,553,151 describes a so-called "forward masking".
  • weak consonants are temporally covered by the previous strong vowels.
  • This publication proposes a relatively fast compressor with an "attack time” of about 10 msec. and a release time of about 75 to 150 msec. in front.
  • a problem in the previously known systems for increasing the speech intelligibility of speech in audio signals is their relatively high complexity, which means that both a high software cost for calculating the individual algorithms and a high hardware cost is necessary.
  • the audio signal is changed so that the language no longer sounds very natural.
  • the speech signal can be disrupted, which can even counteract improved intelligibility.
  • the aim of the present invention is therefore to provide a circuit arrangement for improving the speech quality of audio signals, on the one hand requires little effort and on the other hand, the language still sound natural.
  • the invention is essentially based on amplifying the audio signal to a predetermined factor and filtering it in a high-pass filter, wherein the cutoff frequency of the high-pass filter is controlled such that the amplitude of the audio signal after the processing path is equal to or proportional to the amplitude of the audio signal at the input of the processing path is.
  • the fundamental wave of the speech signal which contributes relatively little to the intelligibility but has the largest energy, can be attenuated and the usual signal spectrum of the audio signal can be correspondingly increased.
  • the amplitude of the vowels (large amplitude, low frequency) in the transition region consonant (small amplitude, high frequency) can be lowered to vowel to reduce the so-called "backward masking".
  • the entire signal is increased by a factor of g. This factor controls the strength of the signal enhancement effect, with meaningful values for the factor g being between about 1.5 and 4.
  • the circuit arrangement according to the invention are raised as higher-frequency components and lowered the low-frequency fundamental wave to the same extent, so that the amplitude (or energy) of the audio signal remains unchanged.
  • the corner frequency of the variable high-pass filter can be lowered with the circuit arrangement according to the present invention. Therefore, an "offset" can be added in the control to the input signal, which is either fixed or proportional to the peak amplitude of the input-side audio signal.
  • the corner frequency f c of the variable high-pass filter is limited downwards, since the lowest frequency for speech is about 200 Hz. A range of approximately 100 to 120 Hz has proven suitable for a lower corner frequency.
  • the circuit arrangement has a variable high pass 20, which is variable in its corner frequency f c .
  • the variable high-pass filter 20 has a control input 21 to which a control signal for changing the corner frequency f c is can be applied.
  • This variable high-pass filter 20 is preferably supplied via a low pass filter 10 to be improved.
  • an input terminal 1 is provided for applying the audio signal.
  • the low pass 10 must not provided, but is advantageous to eliminate signal interference in the audio signal.
  • At the output of the variable high-pass filter 20 sits an amplifier stage 30, which amplifies the output-side signal of the variable high-pass filter 20 by a factor of g.
  • This factor g is adjustable and is preferably between about 1.5 and 4. Once set gain factor is preferably not changed.
  • the entire processing path consisting of variable high pass 20 and amplifier 30 and optional low-pass filter 10 has an output terminal 2, at which the processed audio signal can be tapped as an output signal.
  • a regulation of the cut-off frequency f c of the variable high-pass filter 20 is carried out in the following manner for improving speech intelligibility of speech within the audio signal. If the amplitude (or energy) of the input signal at the input 1 of the circuit arrangement is greater than the amplitude (or energy) at the output 2 of the transmission path, then the corner frequency f c is lowered. Incidentally, increased. Provided that the amplitudes at the input 1 and output 2 are the same or proportional to a predetermined factor, no further change in the corner frequency f c.
  • FIG. 2 shows a development of the circuit arrangement of Figure 1 is shown.
  • FIG. 2 shows a comparator 36 with a downstream integrator, which is preceded by a scaling factor Ki.
  • the output terminal of the integrator 40 is connected to the control input 21 of the variable high-pass filter 20 in combination.
  • the comparator 36 has two input terminals 34, 35, at the first terminal 34, the input signal and at its terminal 35, the output signal of the transmission path is applied.
  • the circuit arrangement of FIG. 3 differs from the circuit arrangement of FIG. 2 in that the integrator 40 is replaced by a digital circuit arrangement 60.
  • the corner frequency f c is increased or decreased by a step d, depending on whether the output signal xc at the output of the comparator 36 is greater or less than 0.
  • FIG. 4 shows a further development of the circuit arrangement according to the invention.
  • the development consists in adding an offset K to the input signal present at the input 34.
  • This offset may be chosen to be constant or may be a factor K weighted output of a peak detector 70.
  • the audio signal is applied on the input side.
  • variable high-pass filter 20 is responsible.
  • the circuit operates as follows: A vowel is low-frequency with large amplitude. A consonant, in contrast, is high-frequency with a small amplitude.
  • the amplification factor g is set so that a gain of 6 dB is achieved. Due to the low-frequency vowel, the corner frequency of the variable high-pass filter 20 has adjusted to this low frequency. The fundamental wave is lowered so far that the Output amplitude is the same input amplitude of the audio signal, although the gain of 6 dB has been selected.
  • the inventive circuit arrangement of FIG. 1 operates as follows.
  • the high pass filter 20 has adjusted to the frequency of the consonant.
  • the amplitude of the input signal corresponds to the amplitude of the output signal.
  • the relatively high cut-off frequency f c of the high-pass filter 20 attenuates the vowels in the temporal transition and consequently does not cover the consonant.
  • the corner frequency f c is adjusted due to the control loop so that the amplitude of the input signal corresponds to the amplitude of the output signal.
  • the audio path (high pass, low pass, gain) is calculated separately for left and right, but the high passes have the same corner frequency f c .

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  • Engineering & Computer Science (AREA)
  • Human Computer Interaction (AREA)
  • Quality & Reliability (AREA)
  • Signal Processing (AREA)
  • Health & Medical Sciences (AREA)
  • Audiology, Speech & Language Pathology (AREA)
  • Computational Linguistics (AREA)
  • Physics & Mathematics (AREA)
  • Acoustics & Sound (AREA)
  • Multimedia (AREA)
  • Tone Control, Compression And Expansion, Limiting Amplitude (AREA)
  • Circuit For Audible Band Transducer (AREA)
  • Stereophonic System (AREA)

Description

Die Erfindung betrifft eine Schaltungsanordnung zur Verbesserung der Verständlichkeit von Sprache enthaltenden Audiosignalen gemäß den Merkmalen des Oberbegriffs des Anspruchs 1.The invention relates to a circuit arrangement for improving the intelligibility of speech-containing audio signals according to the features of the preamble of claim 1.

Eine solche Anordnung ist z.B. aus US-A- 5,083,312 bekannt.Such an arrangement is e.g. from US-A-5,083,312.

Es gibt verschiedene Möglichkeiten, wie die Sprachverständlichkeit von Audiosignalen verbessert werden kann. Eine Möglichkeit liegt in der Verbesserung des verrauschten Signals. Eine andere Möglichkeit liegt darin, solche Signale zu verbessern, die durch Hall und Echos etc. degradiert wurden. Schließlich kann ein gutes Audiosignal verändert werden, so dass es für Schwerhörige besser verständlich wird. Dies wird beispeilsweise mit Hörgeräten erreicht. Letzlich ist die Veränderung eines guten Audiosignals möglich, so dass es bei starken Hintergrundgeräuschen besser verständlich ist.There are several ways in which the speech intelligibility of audio signals can be improved. One possibility is to improve the noisy signal. Another possibility is to improve signals that have been degraded by reverberation and echoes. Finally, a good audio signal can be changed so that it is easier to understand for the hard of hearing. This is achieved, for example, with hearing aids. Finally, the change of a good audio signal is possible, so that it is better understood with strong background noise.

Ziel der vorliegenden Erfindung ist es, die Sprachverständlichkeit eines verhältnismäßig guten Audiosignals bei unveränderter Lautstärke zu verbessern. Dies bedeutet, gleiche Verständlichkeit bei geringerer Lautstärke oder verbesserte Verständlichkeit bei Umgebungslärm.The aim of the present invention is to improve the speech intelligibility of a relatively good audio signal with unchanged volume. This means equal intelligibility at lower volume or improved intelligibility in ambient noise.

Aus US 5,459,813 ist es bekannt, dass sogenannte "unvoiced sounds" (z. B. Konsonanten) von den viel stärkeren "voiced sounds" (z. B. Vokale) überdeckt werden. Da die "unvoiced sounds" wichtig für die Sprachverständlichkeit sind, wird in dieser Veröffentlichung vorgeschlagen, diese z. B. durch Clipping oder Amplitudenkompression zu verstärken.From US 5,459,813 it is known that so-called "unvoiced sounds" (eg consonants) are covered by the much stronger "voiced sounds" (eg vowels). Since the "unvoiced sounds" are important for speech intelligibility, it is proposed in this publication, this z. B. amplify by clipping or amplitude compression.

In der Veröffentlichung "effects of amplitud distorsion upon intellegibility of speech" von J. C. Liqulider in dem Journal of acustical society of america, Oktober 1946 ist ein sogenanntes "peak clipping" bekannt. Ein solches "peak clipping" ohne Umgebungsrauschen hat kaum Einfluss auf die Sprachverständlichkeit. Ein "peak clipping" bei -20 dB führt immer noch zu einer Verständlchkeit von etwa 96%. Das sogenannte "center clipping" ist wesentlich schlechter, da hier die Konsonanten entfernt werden, die für die Verständlichkeit besonders wichtig sind. "Peak clipping" bei -24 dB braucht nur eine Verstärkung von etwa 14 dB, um dieselbe Verständlichkeit zu erreichen. Aus der Veröffentlichung Elwood Kretsinger et al "The Use of fast Limiting to improve the Intelligibility of Speech in Noise", Speech Monographs, March 1960 ist es bekannt, dass Konsonanten ca. 12 dB schwächer als Vokale sind. Verstärkt man die Konsonanten relativ zu den Vokalen, wird deshalb die Verständlichkeit von Sprache im Audiosignal erhöht. Ersetzt man den Clipper durch einen schnellen "peak limitter" (22 msec) kann man die Verständlichkeit noch weiter erhöhen. Bei -10 dBlimitting erhöhte sich die Verständlichkeit von 56 % auf 84 %.In the publication "Effects of amplitude distorsion upon intellegibility of speech" by J. C. Liqulider in the Journal of Acoustic Society of America, October 1946, a so-called "peak clipping" is known. Such a "peak clipping" without ambient noise has hardly any influence on the speech intelligibility. A peak clipping at -20 dB still leads to an understanding of about 96%. The so-called "center clipping" is much worse, since it removes the consonants, which are particularly important for clarity. Peak clipping at -24 dB only needs about 14 dB gain to achieve the same level of intelligibility. From the publication Elwood Kretzinger et al "The Use of Almost Limiting to Improve the Intelligibility of Speech in Noise," Speech Monographs, March 1960, it is known that consonants are about 12 dB weaker than vowels. Increasing the consonants relative to the vowels increases the intelligibility of speech in the audio signal. If you replace the clipper with a fast "peak limiter" (22 msec) you can increase the intelligibility even further. At -10 dBlimitting, intelligibility increased from 56% to 84%.

Aus Veröffentlichung Ian Thomas et al. "The Intelligibility of filtered-clipped Speech in Noise", The Journal of the Audio Engineering Society, June 1970 ist es bekannt, dass die Grundwelle eines Audiosignals, das Sprache enthält, nur wenig zur Sprachverständlichkeit beiträgt, während die erste Resonanzfrequenz sehr wichtig ist. Deshalb sollte das Signal vor dem Clipping hochpassgefiltert werden.From publication Ian Thomas et al. "The Intelligibility of Filtered-Clipped Speech in Noise," June 1970, it is known that the fundamental of an audio signal containing speech contributes little to speech intelligibility, while the first resonant frequency is very important. Therefore, the signal should be high-pass filtered before clipping.

Aus Veröffentlichung Ian Thomas et al., "Intelligibility enhancement through spectral weigthing", Proceedings of the 1972 IEEE Conference on Speech Communication and Processing ist es bekannt, dass das Clipping zwar die Verständlichkeit von Sprache erhöht, jedoch die Signalqualität beeinträchtigt. In dieser Veröffentlichung wird deshalb vorgeschlagen, die Signalenergie in die signifikanten Frequenzbereiche zu verlagern.From publication Ian Thomas et al., "Intelligibility enhancement through spectral weigthing", Proceedings of the 1972 IEEE Conferencing on Speech Communication and Processing, it is well known that while clipping improves speech intelligibility, it compromises signal quality. In this publication it is therefore proposed to shift the signal energy into the significant frequency ranges.

Aus US 5,479,560 ist es darüber hinaus bekannt, das Audiosignal in mehrere Frequenzbänder aufzuteilen und diejenigen Frequenzbänder mit großer Energie verhältnismäßig stark zu verstärken und die anderen abzusenken. Dies wird deshalb vorgeschlagen, weil Sprache aus einer Aneinanderreihung von Phonehmen besteht. Phoneme bestehen aus einer Vielzahl von Frequenzen. Diese werden an den Resonanzfrequenzen des Mund- und Rachenraums besonders verstärkt. Ein Frequenzband mit solche einem spektralen Peak wird Formant genannt. Formants sind besonders wichtig zur Erkennung von Phonemen und somit Sprache. Ein Ansatz zur Verbesserung der Sprachverständlichkeit ist es daher, die Peaks (Formants) des Frequenzspektrums eines Audiosignals zu verstärken und die dazwischen liegenden Täler abzuschwächen. Für einen Erwachsenen Mann liegt die Grundfrequenz von Sprache bei etwa 60 bis 250 Hz. Die ersten vier Formants liegen bei 500 Hz, 1 500 Hz, 2 500 Hz und 3 500 Hz (vgl. hierzu US-Patent 5,459,813.From US 5,479,560 it is also known to divide the audio signal into several frequency bands and relatively strong amplify those frequency bands with high energy and lower the other. This is suggested because language consists of a sequence of phonemes. Phones are made up of a variety of frequencies. These are particularly enhanced at the resonance frequencies of the mouth and throat area. A frequency band with such a spectral peak is called a formant. Formants are particularly important for recognizing phonemes and thus speech. One approach to improving speech intelligibility is therefore to amplify the peaks (formants) of the frequency spectrum of an audio signal and to attenuate the intervening valleys. For an adult male, the fundamental frequency of speech is about 60 to 250 Hz. The first four formants are at 500 Hz, 1500 Hz, 2500 Hz, and 3500 Hz (see US Patent 5,459,813.

Aus US 4,454,609 ist es bekannt, hauptsächlich die Konsonanten zu verstärken.From US 4,454,609 it is known to amplify mainly the consonants.

Schließlich beschreibt US 5,553,151 ein sogenanntes "forward masking". Hierbei werden schwache Konsonanten durch die vorhergehenden starken Vokale zeitlich überdeckt. Diese Veröffentlichung schlägt einen verhältnismäßig schnellen Kompressor mit einer "attack time" von ca. 10 msec. und einer "release time" von ca. 75 bis 150 msec. vor.Finally, US 5,553,151 describes a so-called "forward masking". Here, weak consonants are temporally covered by the previous strong vowels. This publication proposes a relatively fast compressor with an "attack time" of about 10 msec. and a release time of about 75 to 150 msec. in front.

Problematisch bei den bisher bekannten Systemen zur Erhöhung der Sprachverständlichkeit von Sprache in Audiosignalen ist deren verhältnismäßig hohe Komplexität, das bedeutet, dass sowohl ein hoher Softwareaufwand zur Berechnung der einzelnen Allgorithmen sowie ein hoher Hardwareaufwand notwendig ist. Bei einfacheren Systemen wird dagegen das Audiosignal so verändert, dass die Sprache nicht mehr sehr natürlich klingt. Des Weiteren kann bei einfachen Systemen dem Sprachsignal Störungen zugefügt werden, das einer verbesserten Verständlichkeit sogar entgegen wirken kann.A problem in the previously known systems for increasing the speech intelligibility of speech in audio signals is their relatively high complexity, which means that both a high software cost for calculating the individual algorithms and a high hardware cost is necessary. In simpler systems, however, the audio signal is changed so that the language no longer sounds very natural. Furthermore, in simple systems, the speech signal can be disrupted, which can even counteract improved intelligibility.

Ziel der vorliegenden Erfindung ist es daher, eine Schaltungsanordnung zur Verbesserung der Sprachqualität von Audiosignalen anzugeben, das einerseits geringen Aufwand erfordert und andererseits die Sprache noch natürlich klingen lässt.The aim of the present invention is therefore to provide a circuit arrangement for improving the speech quality of audio signals, on the one hand requires little effort and on the other hand, the language still sound natural.

Dieses Ziel wird durch eine Schaltungsanordnung mit dem Merkmale des Anspruchs 1 gelöst.This object is achieved by a circuit arrangement with the features of claim 1.

Weiterbildungen einer solchen Schaltungsanordnung sind Gegenstand der Unteransprüche.Further developments of such a circuit arrangement are the subject of the dependent claims.

Die Erfindung beruht im Wesentlichen darauf, das Audiosignal auf einen vorgegebenen Faktor zu verstärken und in einem Hochpass zu filtern, wobei die Eckfrequenz des Hochpasses so geregelt wird, dass die Amplitude des Audiosignals nach der Verarbeitungsstrecke gleich oder proportional der Amplitude des Audiosignals am Eingang der Verarbeitungsstrecke ist.The invention is essentially based on amplifying the audio signal to a predetermined factor and filtering it in a high-pass filter, wherein the cutoff frequency of the high-pass filter is controlled such that the amplitude of the audio signal after the processing path is equal to or proportional to the amplitude of the audio signal at the input of the processing path is.

Mit dieser Schaltungsanordnung kann die Grundwelle des Sprachsignals, die relativ wenig zur Verständlichkeit beiträgt, aber die größte Energie besitzt, abgeschwächt werden und das übliche Signalspektrums des Audiosignals entsprechend angehoben werden. Außerdem kann die Amplitude der Vokale (große Amplitude, tiefe Frequenz) im Übergangsbereich Konsonant (kleine Amplitude, große Frequenz) zu Vokal abgesenkt werden, um das sogenannte "backward masking" zu verringern. Dazu wird das gesamte Signal um einen Faktor g angehoben. Dieser Faktor steuert die Stärke des Effekts der Signalverbesserung, wobei sinnvolle Werte für den Faktor g etwa zwischen 1,5 und 4 liegen. Mit der erfindungsgemäßen Schaltungsanordnung werden als höher frequente Anteile angehoben und die tieffrequente Grundwelle im gleichen Maße abgesenkt, so dass die Amplitude (oder Energie) des Audiosignales unverändert bleibt. Für Signalanteile mit kleinen Amplituden, also Konsonanten, kann mit der Schaltungsanordnung nach der vorliegenden Erfindung die Eckfrequenz des variablen Hochpasses abgesenkt werden. Deshalb kann in der Regelung zu dem Eingangssignal noch ein "offset" addiert werden, der entweder fix oder proportional zur Peak-Amplitude des eingangsseitigen Audiosignal ist.With this circuit arrangement, the fundamental wave of the speech signal, which contributes relatively little to the intelligibility but has the largest energy, can be attenuated and the usual signal spectrum of the audio signal can be correspondingly increased. In addition, the amplitude of the vowels (large amplitude, low frequency) in the transition region consonant (small amplitude, high frequency) can be lowered to vowel to reduce the so-called "backward masking". For this purpose, the entire signal is increased by a factor of g. This factor controls the strength of the signal enhancement effect, with meaningful values for the factor g being between about 1.5 and 4. With the circuit arrangement according to the invention are raised as higher-frequency components and lowered the low-frequency fundamental wave to the same extent, so that the amplitude (or energy) of the audio signal remains unchanged. For signal components with small amplitudes, that is to say consonants, the corner frequency of the variable high-pass filter can be lowered with the circuit arrangement according to the present invention. Therefore, an "offset" can be added in the control to the input signal, which is either fixed or proportional to the peak amplitude of the input-side audio signal.

In einer Weiterbildung der Erfindung ist vorgesehen, dass höherfrequenzte Signalanteile im Audiosignal abgesenkt werden. Mit einem Tiefpass vor dem variablen Hochpass können Störungen im Signal unterdrückt werden.In a development of the invention, it is provided that higher-frequency signal components in the audio signal are lowered. With a low pass in front of the variable highpass, disturbances in the signal can be suppressed.

In einer Weiterbildung der Erfindung ist vorgesehen, dass die Eckfrequenz fc des variablen Hochpassfilters nach unten begrenzt wird, da die unterste Frequenz für Sprache bei ca. 200 Hz liegt. Bewährt hat sich für eine untere Eckfrequenz ein Bereich von etwa 100 bis 120 Hz.In one embodiment of the invention, it is provided that the corner frequency f c of the variable high-pass filter is limited downwards, since the lowest frequency for speech is about 200 Hz. A range of approximately 100 to 120 Hz has proven suitable for a lower corner frequency.

Nachfolgend wird die erfindungsgemäße Schaltungsanordnung anhand von Figuren beispielhaft erläutert. Es zeigen:

Figur 1
die prinzipielle Schaltungsanordnung zur Verbesserung der Sprachverständlichkeit in einem Audiosignal,
Figur 2
eine Weiterbildung der Schaltungsanordnung von Figur 1,
Figur 3
eine andere Weiterbildung der Schaltungsanordnung von Figur 1, und
Figur 4
eine andere Weiterbildung der Schaltungsanordnung von Figur 1, und
Figur 5
eine vierte Weiterbildung der erfindungsgemäßen Schaltungsanordnung.
The circuit arrangement according to the invention is explained by way of example with reference to figures. Show it:
FIG. 1
the basic circuit arrangement for improving speech intelligibility in an audio signal,
FIG. 2
1 development of the circuit arrangement of FIG. 1,
FIG. 3
another development of the circuit arrangement of Figure 1, and
FIG. 4
another development of the circuit arrangement of Figure 1, and
FIG. 5
A fourth development of the circuit arrangement according to the invention.

In den nachfolgenden Figuren bezeichnen gleiche Bezugszeichen, sofern nicht anders angegeben, gleiche Teile mit gleicher Bedeutung.In the following figures, like reference numerals, unless otherwise indicated, like parts with the same meaning.

In Figur 1 ist der prinzipielle Aufbau der erfindungsgemäßen Schaltungsanordnung gezeigt. Die Schaltungsanordnung weist einen variablen Hochpass 20 auf, der in seiner Eckfrequenz fc veränderbar ist. Hierfür verfügt der variable Hochpass 20 über einen Steuereingang 21, an dem ein Steuersignal zur Veränderung der Eckfrequenz fc anlegbar ist. Diesem variablen Hochpass 20 wird vorzugsweise über einen Tiefpass 10 das zu verbessernde Audiosignal zugeführt. Hierfür ist eine Eingangsklemme 1 zum Anlegen des Audiosignals vorgesehen. Der Tiefpass 10 muss nicht vorgesehen sein, ist jedoch vorteilhaft, um Signalstörungen im Audiosignal zu beseitigen. Am Ausgang des variablen Hochpasses 20 sitzt eine Verstärkerstufe 30, die das ausgangsseitige Signal des variablen Hochpasses 20 um einen Faktor g verstärkt. Dieser Faktor g ist einstellbar und liegt vorzugsweise zwischen etwa 1,5 und 4. Ein einmal eingstellter Verstärkungsfaktor wird vorzugsweise nicht mehr verändert. Die gesamte Verarbeitungsstrecke bestehend aus variablen Hochpass 20 und Verstärker 30 sowie optionalem Tiefpass 10 verfügt über eine Ausgangsklemme 2, an der das verarbeitete Audiosignal als Ausgangssignal abgreifbar ist.In Figure 1, the basic structure of the circuit arrangement according to the invention is shown. The circuit arrangement has a variable high pass 20, which is variable in its corner frequency f c . For this purpose, the variable high-pass filter 20 has a control input 21 to which a control signal for changing the corner frequency f c is can be applied. This variable high-pass filter 20 is preferably supplied via a low pass filter 10 to be improved. For this purpose, an input terminal 1 is provided for applying the audio signal. The low pass 10 must not provided, but is advantageous to eliminate signal interference in the audio signal. At the output of the variable high-pass filter 20 sits an amplifier stage 30, which amplifies the output-side signal of the variable high-pass filter 20 by a factor of g. This factor g is adjustable and is preferably between about 1.5 and 4. Once set gain factor is preferably not changed. The entire processing path consisting of variable high pass 20 and amplifier 30 and optional low-pass filter 10 has an output terminal 2, at which the processed audio signal can be tapped as an output signal.

Erfindungsgemäß wird eine Regelung der Eckfrequenz fc des variablen Hochpasses 20 in folgender Art und Weise zur Verbesserung der Sprachverständlichkeit von Sprache innerhalb des Audiosignals durchgeführt. Ist die Amplitude (oder auch Energie) des Eingangssignals am Eingang 1 der Schaltungsanordnung größer als die Amplitude (oder Energie) am Ausgang 2 der Übertragungsstrecke, dann wird die Eckfrequenz fc erniedrigt. Im Übrigen erhöht. Sofern die Amplituden am Eingang 1 und Ausgang 2 gleich oder zu einem vorgegebenen Faktor proportional sind, erfolgt keine weitere Veränderung der Eckfrequenz fc.According to the invention, a regulation of the cut-off frequency f c of the variable high-pass filter 20 is carried out in the following manner for improving speech intelligibility of speech within the audio signal. If the amplitude (or energy) of the input signal at the input 1 of the circuit arrangement is greater than the amplitude (or energy) at the output 2 of the transmission path, then the corner frequency f c is lowered. Incidentally, increased. Provided that the amplitudes at the input 1 and output 2 are the same or proportional to a predetermined factor, no further change in the corner frequency f c.

In Figur 2 ist eine Weiterbildung der Schaltungsanordnung von Figur 1 dargestellt. In Figur 2 ist ein Vergleicher 36 mit nachgeschaltetem Integrator, dem ein Skalierungsfaktor Ki vorgeschaltet ist, vorgesehen. Die Ausgangsklemme des Integrators 40 ist mit dem Steuereingang 21 des variablen Hochpasses 20 in Verbindung. Der Vergleicher 36 weist zwei Eingangsklemmen 34, 35 auf, an deren erste Klemme 34 das Eingangssignal und an dessen Klemme 35 das Ausgangssignal der Übertragungsstrecke angelegt wird.2 shows a development of the circuit arrangement of Figure 1 is shown. FIG. 2 shows a comparator 36 with a downstream integrator, which is preceded by a scaling factor Ki. The output terminal of the integrator 40 is connected to the control input 21 of the variable high-pass filter 20 in combination. The comparator 36 has two input terminals 34, 35, at the first terminal 34, the input signal and at its terminal 35, the output signal of the transmission path is applied.

Die Schaltungsanordnung von Figur 3 unterscheidet sich von der Schaltungsanordnung von Figur 2 dadurch, dass der Integrator 40 durch eine digitale Schaltungsanordnung 60 ersetzt ist. In der digitalen Schaltungsanordnung 60 wird nach Maßgabe des Ausgangssignals des Vergleichers 36 die Eckfrequenz fc um einen Schritt d erhöht oder erniedrigt, je nachdem, ob das Ausgangssignal xc am Ausgang des Vergleichers 36 größer oder kleiner 0 ist.The circuit arrangement of FIG. 3 differs from the circuit arrangement of FIG. 2 in that the integrator 40 is replaced by a digital circuit arrangement 60. In the digital circuit arrangement 60, according to the output signal of the comparator 36, the corner frequency f c is increased or decreased by a step d, depending on whether the output signal xc at the output of the comparator 36 is greater or less than 0.

Schließlich ist in Figur 4 noch eine Weiterbildung der erfindungsgemäßen Schaltungsanordnung dargestellt. Die Weiterbildung besteht darin, dass zu dem an dem Eingang 34 anstehenden Eingangssignal ein Offset K addiert wird. Dieser Offset kann konstant gewählt werden oder ein mit einem Faktor K gewichteter Ausgang eines Peak-Detektor 70 sein. An dem Peak-Detektor 70 wird eingangsseitig das Audiosignal angelegt.Finally, FIG. 4 shows a further development of the circuit arrangement according to the invention. The development consists in adding an offset K to the input signal present at the input 34. This offset may be chosen to be constant or may be a factor K weighted output of a peak detector 70. At the peak detector 70, the audio signal is applied on the input side.

Mit der erfindungsgemäßen Schaltungsanordnung gemäß den Figuren 1 bis 4 ist es möglich, die Grundwelle des Audiosignals abzusenken und den restlichen Signalanteil anzuheben. Hierfür ist das variable Hochpassfilter 20 verantwortlich.With the circuit arrangement according to the invention according to the figures 1 to 4, it is possible to lower the fundamental wave of the audio signal and to raise the remaining signal component. For this purpose, the variable high-pass filter 20 is responsible.

Für den Fall, dass im Sprachsignal ein Konsonant einem Vokal folgt, arbeitet die Schaltungsanordnung folgendermaßen: Ein Vokal ist tieffrequent mit großer Amplitude. Ein Konsonant ist dagegen hochfrequent mit kleine Amplitude. Bei der erfindungsgemäßen Schaltungsanordnung wird der Verstärkungsfaktor g so eingestellt, dass eine Verstärkung von 6 dB erreicht wird. Durch den tieffrequenten Vokal hat sich die Eckfrequenz des variablen Hochpassfilters 20 auf diese tiefe Frequenz eingestellt. Die Grundwelle ist also so weit abgesenkt, dass die Ausgangsamplitude gleicher Eingangsamplitude des Audiosignals ist, obwohl die Verstärkung von 6 dB gewählt wurde. Folgt auf den Vokal nun ein Konsonant (höhere Frequenz!) wird dieser sofort um 6 dB angehoben, da die Eckfrequenz des Hochpassfilters 20 noch auf die tiefe Frequenz des Vokals eingestellt ist. Der Konsonant wird als weniger stark vom Vokal überdeckt. Erst nach einigen Millisekunden erhöht sich die Eckfrequenz fc und senkt somit auch den Konsonant ab, so dass die Amplitude des Eingangssignals gleich der Amplitude des Ausgangssignals der Verarbeitungsstrecke ist.In the event that a consonant follows a vowel in the speech signal, the circuit operates as follows: A vowel is low-frequency with large amplitude. A consonant, in contrast, is high-frequency with a small amplitude. In the circuit arrangement according to the invention, the amplification factor g is set so that a gain of 6 dB is achieved. Due to the low-frequency vowel, the corner frequency of the variable high-pass filter 20 has adjusted to this low frequency. The fundamental wave is lowered so far that the Output amplitude is the same input amplitude of the audio signal, although the gain of 6 dB has been selected. If a consonant (higher frequency!) Now follows the vowel, it is immediately raised by 6 dB, because the cut-off frequency of the high-pass filter 20 is still set to the low frequency of the vowel. The consonant is less than covered by the vowel. Only after a few milliseconds, the corner frequency f c increases and thus also decreases the consonant, so that the amplitude of the input signal is equal to the amplitude of the output signal of the processing path.

Bei einem Übergang Konsonant auf Vokal arbeitet die erfindungsgemäße Schaltungsanordnung von Figur 1 folgendermaßen. Das Hochpassfilter 20 hat sich auf die Frequenz des Konsonants eingestellt. Die Amplitude des Eingangssignals entspricht der Amplitude des Ausgangssignals. Folgt nun ein Vokal (tieffrequent) wird durch die verhältnismäßig hohe Eckfrequenz fc des Hochpassfilters 20 der Vokal beim zeitlichen Übergang gedämpft und der Konsonant folglich nicht überdeckt. Erst nach einigen Millisekunden ist die Eckfrequenz fc aufgrund der Regelzeit der Regelschleife so eingeregelt, dass die Amplitude des Eingangssignals der Amplitude des Ausgangssignals entspricht.In a transition from consonant to vowel, the inventive circuit arrangement of FIG. 1 operates as follows. The high pass filter 20 has adjusted to the frequency of the consonant. The amplitude of the input signal corresponds to the amplitude of the output signal. If a vowel follows (low-frequency), the relatively high cut-off frequency f c of the high-pass filter 20 attenuates the vowels in the temporal transition and consequently does not cover the consonant. Only after a few milliseconds, the corner frequency f c is adjusted due to the control loop so that the amplitude of the input signal corresponds to the amplitude of the output signal.

Abschließend ist noch folgendes anzumerken: Bei einem Stereosignal kann entweder jeder Kanal eine eigene Regelung erhalten wie oben beschrieben oder sie können eine gemeinsame Regelung benutzen. Dann ist z. B. (vgl. Figur 5) an den Eingang 34=Abs (Input_Left)+Abs(Input_Right) anzulegen und an den Eingang 35=Abs (Output_Left)+Abs (Output_Right). Der Audiopfad (Hochpass, Tiefpass, Gain) wird für links und rechts getrennt berechnet, die Hochpässe besitzen aber dieselbe Eckfrequenz fc.Finally, the following should be noted: In the case of a stereo signal, either each channel can have its own control as described above, or they can use a common control. Then z. 5) to the input 34 = Abs (Input_Left) + Abs (Input_Right) and to the input 35 = Abs (Output_Left) + Abs (Output_Right). The audio path (high pass, low pass, gain) is calculated separately for left and right, but the high passes have the same corner frequency f c .

Claims (13)

  1. Circuit arrangement for improving the intelligibility of audio signals containing speech, in which frequency and/or amplitude components of the audio signal are changed according to predefined parameters, and the audio signal is passed through a high-pass filter (20) in a processing segment, the corner frequency fc of the high-pass filter being adjustable, and wherein the signal is amplified by a predefinable factor g after the high-pass filter (20), characterised in that, in order to adjust the corner frequency fc of the high-pass filter (20), the amplitude of both the input signal and the output signal of the processing segment is detected, and in that the corner frequency fc is reduced if the amplitude of the input signal is greater than the amplitude of the output signal at the output of the processing segment, and otherwise is increased.
  2. Circuit arrangement according to Claim 1, characterised in that the factor g is selected such that it is > = 1.
  3. Circuit arrangement according to Claim 1 or 2, characterised in that the factor g is selected such that it is approximately between 1.5 and 4.
  4. Circuit arrangement according to one of Claims 1 to 3, characterised in that the change in the corner frequency fc proceeds incrementally, preferably in 1 Hz steps.
  5. Circuit arrangement according to one of Claims 1 to 4, characterised in that the corner frequency fc is variable within the range from approximately 100 Hz to 1 kHz.
  6. Circuit arrangement according to one of Claims 1 to 5, characterised in that the lower corner frequency fc is approximately 100 to 120 Hz.
  7. Circuit arrangement according to one of Claims 1 to 6, characterised in that a low-pass filter (10) is connected upstream of the variable high-pass filter (20).
  8. Circuit arrangement according to Claim 7, characterised in that the low-pass filter (10) has a corner frequency of approximately 6 kHz.
  9. Circuit arrangement according to one of Claims 1 to 8, characterised in that a comparator (36) is connected to one control input (21) of the variable high-pass filter (20) for changing the corner frequency fc, the input signal of the processing segment being applied to one input (34) of the comparator and the output signal of the processing segment being applied to the other input (35) of the comparator.
  10. Circuit arrangement according to Claim 9, characterised in that an integrator (40) is connected between the control input (21) of the variable high-pass filter (20) and the output of the comparator (36).
  11. Circuit arrangement according to Claim 9, characterised in that a digital circuit arrangement (60) for incrementing the corner frequency fc in steps (d) is provided between the control input (21) of the variable high-pass filter (20) and the output of the comparator (36).
  12. Circuit arrangement according to one of Claims 9 to 11, characterised in that an offset is added to the input signal at one input (34) of the comparator (36).
  13. Circuit arrangement according to one of Claims 9 to 12, characterised in that the audio signal is a stereo signal, and in that the sum of the input signals for the left and right channel is fed to a first input (34) of the comparator (36), and in that the sum of the output signal for the left and right channel is fed to the second input (35) of the comparator (36).
EP02005495A 2001-05-18 2002-03-11 Device for improving the intelligibility of audio signals containing speech Expired - Fee Related EP1258865B1 (en)

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DE10124699A DE10124699C1 (en) 2001-05-18 2001-05-18 Circuit arrangement for improving the intelligibility of speech-containing audio signals

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EP1258865A3 (en) 2004-05-06
EP1258865A2 (en) 2002-11-20
US20020173950A1 (en) 2002-11-21
JP2003018691A (en) 2003-01-17
JP4141736B2 (en) 2008-08-27
DE10124699C1 (en) 2002-12-19
US7418379B2 (en) 2008-08-26

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