EP1103956B1 - Réduction exponentielle de bruit et d'écho pendant les pauses de la parole - Google Patents

Réduction exponentielle de bruit et d'écho pendant les pauses de la parole Download PDF

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Publication number
EP1103956B1
EP1103956B1 EP00124577A EP00124577A EP1103956B1 EP 1103956 B1 EP1103956 B1 EP 1103956B1 EP 00124577 A EP00124577 A EP 00124577A EP 00124577 A EP00124577 A EP 00124577A EP 1103956 B1 EP1103956 B1 EP 1103956B1
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European Patent Office
Prior art keywords
signal
noise
speech
signals
echo
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Expired - Lifetime
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EP00124577A
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German (de)
English (en)
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EP1103956A3 (fr
EP1103956A2 (fr
Inventor
Hans Jürgen Matt
Michael Walker
Michael Maurer
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Alcatel CIT SA
Alcatel Lucent SAS
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Alcatel CIT SA
Alcatel SA
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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
    • G10L21/0208—Noise filtering
    • 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
    • G10L19/012—Comfort noise or silence coding
    • 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
    • G10L2021/02082—Noise filtering the noise being echo, reverberation of the speech
    • 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
    • G10L2021/02168—Noise filtering characterised by the method used for estimating noise the estimation exclusively taking place during speech pauses

Definitions

  • TK telecommunications
  • so-called echoes occur in telecommunications connections in two-wire TK networks as line echoes and for example in simple and more uncomfortable TK terminals in the form of acoustic echoes occur.
  • the noise in the first Speech pauses measured and in the form of a power density spectrum continuously stored in a memory.
  • the power density spectrum is over won a Fourier transformation.
  • the stored noise spectrum "as the best current estimate” Subtracted from the current disturbed speech spectrum, then in the Time domain transformed back to this way a noise reduction for the disturbed signal.
  • the power density spectra for the noise and for the speech itself are first estimated by means of a spectral subtraction.
  • a spectral acoustic masking threshold R T (f) for the human ear is then calculated using the rules from the MPEG standard, for example.
  • a filter pass curve H (f) is then calculated according to a simple rule, which is designed so that essential spectral parts of the speech are transmitted as unchanged as possible and spectral parts of the noise are lowered as far as possible.
  • the compander initially has the property of having speech signals certain (pre-set) "normal speech signal level" (possibly normal Called volume) virtually unchanged from its entrance to the exit transferred to.
  • the compander thus consists of a compressor for speech signal levels, which are greater than or equal to a normal level and an expander for Signal levels that are less than the normal level.
  • the gain reduction in the expander becomes stronger with increasingly smaller input levels.
  • Object of the present invention is in contrast, a method with to present the features described above, in which possible inexpensive and inexpensive way without big Computing and with low demand for computer memory and Data storage is an echo and noise reduction is effected with simple means as pleasant as possible for the human ear Acoustic overall impression produced, depending on the taste in addition to individual needs can be adjusted.
  • a significant noise reduction is achieved.
  • that a reduction of the noise during a speech break the hearing significantly less burdened by making the deafness effect louder Sound effect significantly reduces.
  • the ear can be reinserted when the Speech more responsive and listen more carefully.
  • the value of the time constant ⁇ 1 in humans is usually between 50 ms to 150 ms and is preferably about 65 ms.
  • a 0 (k + 1) assumes a predetermined constant value c 2 during a speech pause and / or a value c 3 ⁇ c 2 if an echo signal is present if the predecessor value a 0 (k ) ⁇ c 2 has become.
  • the given function f (N) is a function g (S / N), which is derived from the quotient S / N from the power value of the signal level S the useful signals to be transmitted and the power value of the noise level N or that the given function f (N) is a function g '(N / S), which depends on the reciprocal N / S of this quotient.
  • the given function f (N) is a function g '(N / S), which depends on the reciprocal N / S of this quotient.
  • noise reduction f max or g max should be at a maximum between 20 and 30, preferably about 25 dB.
  • the functions f (N) or g (S / N) or g '(N / S) are chosen so that the Reduction of the noise level N is correct according to the psychoacoustic Means of the human auditory spectrum is done.
  • the value for S and / or N not only from the instantaneous power value alone, but also determined from a weighted spectral curve of S and N and in total, a hearing aid, i. a psychoacoustically pleasant sounding noise reduction achieved. Since there is no easily representable measure for an acoustically pleasant sounding Noise reduction is there, all quality assessments are extensive Hearing tests dependent, which then optimized by means of statistical Methods are evaluated to a rating scale, (similar to Speech codecs).
  • a good noise level estimation requires a good speech pause detector, because one can only be sure that in the speech pause sections just annoying noise and not some mix between noise and speech fragments, as is common in practice occurs.
  • a method variant which is characterized by this is particularly preferred characterized in that in the speech pause detector from the input signal x a short-term output sam (x) by means of a short-term level estimator, by means of a middle-time level estimator, a middle-time output signal mam (x) and by means of a long-term level estimator, a long-term output signal lam (x) is formed, that the three output signals sam (x), mam (x) and lam (x) via suitable gain coefficients are set to be approximately equal large, if the input signal x is a pure noise signal, where sam (x) ⁇ mam (x) ⁇ lam (x), that the three output signals sam (x), mam (x) and lam (x) be monitored by comparators, and that the presence of a Speech signal is assumed as the input signal x, if sam (x) and mam (x) initially each become larger than lam (x), and the presence of a Speech signal is
  • a development of this variant of the method provides that the three output signals sam (x), mam (x) and lam (x) for speech pause estimation be supplied to a neural network, which with a variety was trained by scenarios with different input signals x.
  • One Neural network may advantageously have linear and non-linear relationships between a large amount of input parameters and the desired ones Map output values.
  • a prerequisite for this is that the neural Net once with a sufficient amount of input values and training associated output values. Therefore, neural are suitable Networks especially for the task of speech pause detection Presence of different disturbing noises.
  • the presence of echo signals is also detected and / or predicted, and the corresponding echo signals are suppressed or reduced.
  • echoes also occur in a telephone channel, they can generally be predicted on the basis of a previously determined signal propagation time ⁇ E of an echo and the previously determined echo coupling ERL in the channel and the signal strength ES which triggers the echo in the return channel. This estimation can be carried out in such a way that the magnitude of the delayed incoming echoes is estimated as a function of the transmitted speech signal and its instantaneous power.
  • this echo-loaded signal is preferably additionally briefly damped, for example by the abovementioned exponential depression, to a value which is necessary for a substantial reduction of the echo signal.
  • a compander characteristic curve can also be briefly moved in the direction of greater input volume for echoes and returned to its original position after the echoes have faded away.
  • R which describes a reduction of signal levels for both noises and echoes: R (S, N, ES, ⁇ e , ERL, thrs) ⁇ g (S / N) d (ES, ⁇ e , ERL, thrs), where g (S / N) is the above-described noise reduction and dummy is the independently additionally occurring echo reduction when the estimated echo signal exceeds the predetermined threshold value thrs.
  • a noise reduction is also at a constant noise level constant.
  • a sudden onset of echo reduction in the rhythm of Language also means a noise reduction (at least in the short Period) in the speech rhythm. This leads to a pulsed Background noise, which does not sound natural. Therefore, it is advantageous, in the moments of an additional echo reduction Synthetic noise of a suitable noise generator in the Magnitude of the normal background noise to the processed signal to add. This should be as consistent as possible background noise be conveyed to the listener.
  • spectral subtraction with Downstream level reduction in the speech pauses is that First, by means of spectral subtraction, a part of the noise from the Speech signal itself is eliminated and only then the speech pauses in the described type of noise and echoes are released. Total results This combination in subjective tests better hearing impressions than just one simple spectral subtraction.
  • the method provides that the useful signal to be transmitted is a subjected to human hearing adapted spectral filtering.
  • the means of a spectral subtraction first a Estimation of noises, speech and echoes performed, then determines an auricular masking threshold and then the entire Signal via a suitably set transmission filter so edited that the Language components as unadulterated as possible and the echo and noise components be suppressed as much as possible.
  • a combination with the downstream level reduction in the Speech pauses further improve the listening experience.
  • no (or in other examples only a small) suppression of interference signals in the overall signal is carried out, so that the speech signal is forwarded as unadulterated as possible and unhindered.
  • FIG. 2 schematically shows the mode of operation of an arrangement for noise and echo reduction in accordance with the abovementioned reduction function R (S, N, ES, ⁇ E , ERL, thrs) with a speech pause detector SPD.
  • the function value g or g 'for the case S / N ⁇ 0 dB, ie for an extremely high background noise changes into a constant value g 0 of the noise reduction of approximately 6 dB.
  • S / N 0 dB
  • an increased noise reduction is achieved with increasing improvement of the signal-to-noise ratio S / N, reaching a maximum g max ⁇ 25 dB at approximately S / N ⁇ 12 dB.
  • S / N the degree of noise reduction finally drops to zero in order to perform as little manipulation in the transmitted useful signal with low background noise.

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  • Engineering & Computer Science (AREA)
  • Human Computer Interaction (AREA)
  • Signal Processing (AREA)
  • Health & Medical Sciences (AREA)
  • Audiology, Speech & Language Pathology (AREA)
  • Computational Linguistics (AREA)
  • Physics & Mathematics (AREA)
  • Acoustics & Sound (AREA)
  • Multimedia (AREA)
  • Quality & Reliability (AREA)
  • Cable Transmission Systems, Equalization Of Radio And Reduction Of Echo (AREA)
  • Telephone Function (AREA)
  • Circuit For Audible Band Transducer (AREA)
  • Soundproofing, Sound Blocking, And Sound Damping (AREA)

Claims (23)

  1. Procédé de réduction des signaux d'écho et/ou de bruit dans les systèmes de télécommunication (=TK) destinés à la transmission de signaux utiles sonores, notamment la voix humaine, avec lequel il est déterminé, au moyen de la détection des pauses de la parole, à quel moment le mélange à transmettre de signaux utiles et de signaux parasites contient un signal de parole ou à quel moment il se produit une pause de la parole, un multiplicateur comprenant deux entrées modifiant l'amplitude des signaux utiles généralement perturbés par des signaux d'écho et/ou de bruit par un signal de commande dépendant du temps a0(t) ou par un signal de commande a0(k) cadencé au rythme d'un taux d'échantillonnage fT = 1/T, k ε N comptant les valeurs échantillonnées et T désignant la durée de la période entre une valeur échantillonnée et la suivante
       caractérisé en ce    que le signal de commande a0(t) ou a0(k) subit une variation de telle sorte que pendant la présence de signaux vocaux dans le signal utile, l'amplitude du signal de commande a0(t) ou a0(k) est fixée à un valeur constante donnée c0 et, au début d'une pause de la parole dans le signal utile, l'amplitude du signal de commande a0(t) ou a0(k) est continuellement réduite entre une valeur échantillonnée et la suivante d'après la formule de récurrence a0(k+1) = a0(k).β avec β < 1    et que a0(k) et amené à la valeur c0 après la fin d'une pause de la parole.
  2. Procédé selon la revendication 1, caractérisé en ce que le facteur β est déterminé à partir du taux d'échantillonnage fT, d'une constante de temps τ1 et d'un pré-facteur c1 constant prédéfini d'après l'équation β = c1.exp(-1/τ1fT).
  3. Procédé selon la revendication 2, caractérisé en ce que la constante de temps τ1 est choisie entre 50 ms et 150 ms, de préférence τ1 ≈ 65 ms.
  4. Procédé selon l'une des revendications précédentes, caractérisé en ce que la valeur constante choisie est c0 = 1.
  5. Procédé selon l'une des revendications précédentes, caractérisé en ce que a0(k+1) prend une valeur constante prédéfinie c2 pendant une pause de la parole et/ou en présence d'un signal d'écho si la valeur précédente a0(k) était ≤ à c2.
  6. Procédé selon l'une des revendications 1 à 4, caractérisé en ce que pendant une pause de la parole et/ou en présence d'un signal d'écho et pour a0(k) ≤ c2, c2 étant une constante prédéfinie, la valeur de la puissance du niveau sonore N dans le canal TK actuellement utilisé est constamment mesurée et/ou évaluée et que le signal de commande a0(k+1) est réglé continuellement en fonction du niveau de bruit actuel d'après l'équation a0(k+1) = f(N), f(N) étant une fonction prédéfinie de N.
  7. Procédé selon la revendication 6, caractérisé en ce que la fonction f(N) prédéfinie est une fonction g(S/N) qui dépend du quotient S/N entre la valeur de la puissance du niveau du signal S des signaux utiles à transmettre et la valeur de la puissance du niveau sonore N, ou que la fonction f(N) prédéfinie est une fonction g'(S/N) qui dépend de la valeur inverse N/S de ce quotient.
  8. Procédé selon la revendication 7, caractérisé en ce que la fonction f(N) ou g(S/N), avec 1/N « N ou S/N = 0 dB, commence par une valeur constante f0 > 0 ou g0 > 0, croít jusqu'à un maximum fmax ou gmax dans la plage entre N ou S/N - 10 dB à 15 dB, de préférence avec N ou S/N = 12 dB, et chute ensuite à une valeur minimale fmin ou gmin, de préférence à 0 dB, f0 étant supérieure ou égale à 5 dB, g0 inférieure ou égale à 10 dB, f0 étant de préférence supérieure ou égale à 6 dB, g0 inférieure ou égale à 8 dB, et fmax étant supérieure ou égale à 20 dB, gmax inférieure ou égale à 30 dB, fmax et gmax étant de préférence égales à 25 dB.
  9. Procédé selon l'une des revendications 6 à 8, caractérisé en ce que la fonction f(N) ou g(S/N) évolue de manière linéaire avec N ou S/N, au moins partiellement, de préférence dans toutes les sections partielles.
  10. Procédé selon l'une des revendications 6 à 8, caractérisé en ce que la fonction f(N) ou g(S/N) est composée de polynômes et évolue sous la forme d'une cloche asymétrique sur N ou S/N.
  11. Procédé selon l'une des revendications 6 à 10, caractérisé en ce que les fonctions f(N) ou g(S/N) ou g'(S/N) sont choisies de telle sorte que la réduction du niveau sonore N s'effectue fidèlement à l'audition conformément aux valeurs psychoacoustiques moyennes du spectre de l'audition humaine.
  12. Procédé selon l'une des revendications précédentes, caractérisé en ce qu'en plus de la détection et de la réduction des signaux de bruit, on détecte et/ou on anticipe la présence de signaux d'échos et les signaux d'écho sont atténués ou réduits.
  13. Procédé selon la revendication 12 et l'une des revendications 6 à 11, caractérisé en ce que le signal de commande a0(k+1) est réglé continuellement d'après a0(k+1) = h(N, SX, ES, τE, ERL), h(N, SX, ES, τE, ERL) étant une fonction prédéfinie de N, S, du signal utile ES dans le sens inverse d'un correspondant TK qui parle, τE un temps de retard constant du signal d'écho et ERL une constante d'atténuation de l'amplitude du signal d'écho.
  14. Procédé selon la revendication 12, caractérisé en ce que les commandes de réduction des signaux de bruit et de réduction des signaux d'écho s'effectuent séparément.
  15. Procédé selon l'une des revendications 12 à 14, caractérisé en ce qu'un signal de bruit synthétique est ajouté en plus au signal utile pendant la durée d'une réduction de l'écho.
  16. Procédé selon la revendication 15, caractérisé en ce que le signal de bruit synthétique comprend une séquence de signal acoustique qui est perçue comme étant agréable du point de vue psychoacoustique (= comfort noise).
  17. Procédé selon la revendication 15, caractérisé en ce que le signal de bruit synthétique comprend un signal de bruit enregistré précédemment pendant la liaison TK courante.
  18. Procédé selon l'une des revendications précédentes, caractérisé en ce que dans un détecteur de pauses de la parole (SPD), un signal de sortie de courte durée sam(x) est formé au moyen d'un évaluateur de niveau de courte durée, un signal de sortie de durée moyenne mam(x) est formé au moyen d'un évaluateur de niveau de moyenne durée et un signal de sortic de longue durée lam(x) est formé au moyen d'un évaluateur de niveau de longue durée à partir du signal d'entrée x, que les trois signaux de sortie sam(x), mam(x) et lam(x) sont réglés par le biais de coefficients d'amplification appropriés de manière à avoir approximativement la même amplitude lorsque le signal d'entrée x est un signal de bruit pur, sam(x) étant inférieur à mam(x) qui est inférieur à lam(x), que les trois signaux de sortie sam(x), mam(x) et lam(x) sont surveillés par de comparateurs et que la présence d'un signal de parole en tant que signal d'entrée x est supposé lorsque sam(x) et mam(x) sont initialement respectivement supérieurs à lam(x) et la présence d'une pause de la parole lorsque sam(x) et/ou mam(x) devient ensuite de nouveau plus petit que lam(x).
  19. Procédé selon la revendication 18, caractérisé en ce que les trois signaux de sortie sam(x), mam(x) et lam(x) sont acheminés, en vue de l'évaluation des pauses de la parole, à un réseau neuronal qui a été formé avec un grand nombre de scénarios comprenant des signaux d'entrée x différents.
  20. Procédé selon l'une des revendications précédentes, caractérisé en ce que le signal utile à transmettre subit une soustraction spectrale.
  21. Procédé selon l'une des revendications précédentes, caractérisé en ce que le signal utile à transmettre subit un filtrage spectral adapté à l'audition humaine.
  22. Module serveur comprenant des moyens pour exécuter chaque étape individuelle du procédé selon l'une des revendications 1 à 21.
  23. Programme informatique qui présente des instructions pour exécuter chaque étape individuelle du procédé selon l'une des revendications 1 à 21 sur un ordinateur lorsque le programme informatique est chargé sur l'ordinateur.
EP00124577A 1999-11-27 2000-11-10 Réduction exponentielle de bruit et d'écho pendant les pauses de la parole Expired - Lifetime EP1103956B1 (fr)

Applications Claiming Priority (2)

Application Number Priority Date Filing Date Title
DE19957221 1999-11-27
DE19957221A DE19957221A1 (de) 1999-11-27 1999-11-27 Exponentielle Echo- und Geräuschabsenkung in Sprachpausen

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EP1103956A2 EP1103956A2 (fr) 2001-05-30
EP1103956A3 EP1103956A3 (fr) 2001-12-05
EP1103956B1 true EP1103956B1 (fr) 2005-06-08

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US (1) US6999920B1 (fr)
EP (1) EP1103956B1 (fr)
JP (1) JP2001202100A (fr)
KR (1) KR20010051980A (fr)
AT (1) ATE297590T1 (fr)
DE (2) DE19957221A1 (fr)

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JP3152171B2 (ja) * 1997-06-16 2001-04-03 日本電気株式会社 適応フィルタ及びステップサイズ制御方法及びプログラムを記録した記録媒体
US6549587B1 (en) * 1999-09-20 2003-04-15 Broadcom Corporation Voice and data exchange over a packet based network with timing recovery

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KR20010051980A (ko) 2001-06-25
ATE297590T1 (de) 2005-06-15
EP1103956A3 (fr) 2001-12-05
DE50010504D1 (de) 2005-07-14
DE19957221A1 (de) 2001-05-31
US6999920B1 (en) 2006-02-14
EP1103956A2 (fr) 2001-05-30
JP2001202100A (ja) 2001-07-27

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