EP1251493B1 - Procédé pour la réduction du bruit avec fréquence parasite auto-adaptative - Google Patents
Procédé pour la réduction du bruit avec fréquence parasite auto-adaptativeInfo
- Publication number
- EP1251493B1 EP1251493B1 EP02008011A EP02008011A EP1251493B1 EP 1251493 B1 EP1251493 B1 EP 1251493B1 EP 02008011 A EP02008011 A EP 02008011A EP 02008011 A EP02008011 A EP 02008011A EP 1251493 B1 EP1251493 B1 EP 1251493B1
- Authority
- EP
- European Patent Office
- Prior art keywords
- signal
- useful signal
- spectral subtraction
- channel
- signals
- 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.)
- Expired - Lifetime
Links
- 238000000034 method Methods 0.000 title claims description 17
- 230000009467 reduction Effects 0.000 title claims description 7
- 230000003595 spectral effect Effects 0.000 claims abstract description 35
- 230000003044 adaptive effect Effects 0.000 claims description 13
- 238000010586 diagram Methods 0.000 description 5
- 238000003491 array Methods 0.000 description 4
- 230000006978 adaptation Effects 0.000 description 3
- 230000035945 sensitivity Effects 0.000 description 3
- 230000008901 benefit Effects 0.000 description 2
- 230000000694 effects Effects 0.000 description 2
- 238000012805 post-processing Methods 0.000 description 2
- 238000012545 processing Methods 0.000 description 2
- 238000013459 approach Methods 0.000 description 1
- 230000002238 attenuated effect Effects 0.000 description 1
- 230000015572 biosynthetic process Effects 0.000 description 1
- 239000002775 capsule Substances 0.000 description 1
- 238000005314 correlation function Methods 0.000 description 1
- 230000001419 dependent effect Effects 0.000 description 1
- 238000011161 development Methods 0.000 description 1
- 230000018109 developmental process Effects 0.000 description 1
- 230000008030 elimination Effects 0.000 description 1
- 238000003379 elimination reaction Methods 0.000 description 1
- 230000006870 function Effects 0.000 description 1
- 210000003734 kidney Anatomy 0.000 description 1
- 238000004519 manufacturing process Methods 0.000 description 1
- 230000004048 modification Effects 0.000 description 1
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- 238000007493 shaping process Methods 0.000 description 1
- 230000001629 suppression Effects 0.000 description 1
- 230000001052 transient effect Effects 0.000 description 1
Images
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
- 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
- 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
- G10L2021/02161—Number of inputs available containing the signal or the noise to be suppressed
- G10L2021/02165—Two microphones, one receiving mainly the noise signal and the other one mainly the speech signal
Definitions
- the invention relates to a method for noise reduction according to the preamble of patent claim 1.
- a frequently used method for noise reduction of a disturbed useful signal eg a speech signal, music signal etc. is the spectral subtraction.
- Advantage of the spectral subtraction is the low complexity and that the disturbed useful signal is needed only in one variant (only 1 channel).
- Disadvantage is the signal delay (due to the block processing in the spectral range), the limited maximum achievable noise reduction and the difficulty to compensate for transient noises.
- Stationary noise can be reduced, for example, by 12dB with good voice quality.
- microphone arrays are used. Of the various microphone arrays, such are for many practical applications particularly interesting, which make do with small geometrical dimensions for the microphone arrangement.
- Small differential microphone arrays also called super-directive arrays
- LMS least mean square
- two microphones are time-compensated subtracted in two ways such that a virtual microphone with a cardioid polar pattern is turned away from the speaker and a "virtual" microphone with a kidney-shaped characteristic faces away from the speaker.
- the delay compensation corresponds to the time that the sound needs for the distance between the two microphones, eg 1.5cm. This results in a "back-to-back" kidney-shaped directional characteristic.
- the speaker-directed microphone is the primary signal for the adaptive filter, and the oppositely-directed microphone is the reference signal of the interference.
- Figure 1 shows an adaptive arrangement for a beamformer.
- the runtime compensation with an all-pass ALL is realized by shifting by whole samples.
- the combination of two single microphones with omnidirectional characteristic described above results in a cardioid polar pattern characteristic for the speaker and an oppositely directed cardioid polar pattern characteristic as interference reference.
- the adaptive filter H 1 is adapted in the time domain using the LMS (least mean square) algorithm.
- a low-pass filter TP at the system output raises low frequency components, which are attenuated during the formation of the cardioid polar pattern.
- the arrangement of the microphones M in succession according to FIG. 1 is referred to as an "end fire array", in contrast the arrangement of the microphones is designated side by side with a "broadside array”.
- Figure 2 shows an arrangement for a "broad side array" of two microphones in the distance, wherein the two microphone signals are preprocessed with the aid of spectral subtraction (SPS).
- SPS spectral subtraction
- a runtime compensation with the all-pass all between both channels is carried out and serves to compensate for movements of the speaker.
- the sum of the two pre-processed microphone signals forms the primary input and the difference the reference input for an adaptive filter H 1.
- the adaptive filter in this arrangement with sum and difference input is also referred to as 'generalized sidelobe canceller'.
- the adaptation takes place with the LMS algorithm, whereby the implementation of the LMS in the frequency domain takes place.
- a post-processing of the microphone signals is performed with a modified cross-correlation function in the frequency domain.
- the basic structure with spectral preprocessing by means of SPS, beam shaping and post-processing is described in the patent EP 0615226B1, wherein a precise specification of the beamformer has not been made.
- Figure 3 shows an overview of circuitry of microphones for forming the directional characteristics for two microphones.
- the two individual microphones themselves can already have a kidney-shaped characteristic or the so-called spherical characteristics.
- ALL refers to a passpipe all-pass.
- 'Gain' is a gain equalization between both channels which is required in practice to equalize the sensitivity of the microphone capsules.
- the Einschurgicardi in the polar diagrams of the directional characteristics is 90 °.
- the first 3 arrangements a, b and c are suitable as a voice channel, since there is a maximum at 90 ° and an attenuation is present for the other directions.
- Arrangement a and b lead to the same directional characteristic.
- the arrangements a, b are referred to as a sum or difference array and arrangement c as a differential array.
- the arrangements d and e have a zero at 90 ° in the polar diagram and are therefore suitable as a fault reference.
- the zero point at 90 ° in the polar diagram is necessary so that no speech components get into the reference channel. Speech components in the reference channel lead to partial compensation of the language.
- Under ideal conditions, according to arrangements d and e, for the disturbance reference a zero will be set towards the speaker. However, in practical applications this will not be the case. The consequence is that speech components are treated as interference signals and thus removed from the actual speech signal.
- Beamformers are usually adapted only in the speech pauses, in order to allow no adaptation to speech components. Nevertheless, even in this case existing speech components in the reference are compensated since they are always superimposed on the noise.
- the present invention is therefore based on the object to provide a method for noise reduction, with the crosstalk of the useful signal is minmiert in the interference reference signal.
- the invention has the advantage that significantly fewer useful signal components, e.g. Voice components are present in the interference reference signal than with the previous methods.
- the elimination of the disturbing speech components is thus under real conditions with reflections of the speech signal in real spaces such as e.g. in the vehicle possible.
- the invention assumes that a one-sided spectral subtraction is performed to form the interference reference signal. It is essential that the spectral subtraction takes place to form a reference signal only on one channel, which is referred to as 'one-sided'.
- the one channel thus contains useful and interference signals
- the second channel after the spectral subtraction contains only useful signals.
- the useful part is subtracted and the fault remains. This difference is the disturbance reference signal.
- the speech signals are processed so that the interference reference signal has a zero to the speaker in the form of a kidney-shaped or an eight-shaped characteristic.
- the unilateral spectral subtraction leads to a self-regulating control of the characteristic, such that the zero occurs only in voice activity.
- the one-sided spectral subtraction results in nothing or only a small signal being subtracted and thus approximately the characteristic of the single microphone (e.g., kidney or bullet) available for the perturbation.
- the ideal zero for the speech signal in the reference is only achieved with an ideal spectral subtraction in the acoustic free field.
- An ideal spectral subtraction gives the undisturbed speech signal as an output signal and would then make any further processing unnecessary.
- the spectral subtraction in practice gives only a good approximation of the speech signal with noise residues in the speech pauses. Since the one-sided spectral subtraction is used in addition to the microphone zero point, the speech components of the reference reduce significantly.
- the residual noise of the spectral subtraction in speech pauses is set with a parameter, the "spectral floor".
- the spectral floor b is the minimum value of a filter coefficient W of the spectral subtraction at each frequency index i.
- FIG. 4 shows three block diagrams with one-sided spectral subtraction for the reference input.
- the primary useful signal P of the beamformer eg voice signal
- FIG. 4 a the primary useful signal P of the beamformer (eg voice signal) is connected as a differential array DA for the channels 1, 2 (arrangement c in FIG. 3).
- Figure 4b, 4c shows a circuit of the primary signal P as a sum and difference array SD (arrangement a and b in Figure 3).
- the interference reference input processes the reference signal R with the additional extension of the one-sided spectral subtraction in differential form according to the arrangement d and e in FIG. 3.
- the difference between useful signal in channel 2 and interference-canceled useful signal from channel 1 is applied to the adaptive filter H 1.
- the adaptive filter H1 is adapted in the time domain or in an equivalent form in the frequency domain using the LMS algorithm.
- the filtered interference reference signal R is then subtracted from the primary useful signal P.
- a further embodiment of the invention according to Figure 5 is that the one-sided spectral subtraction, PLC 1 'is performed once on the channel 1 for the useful signal to form together with the useful signal in channel 2, a first reference signal R1.
- the unilateral spectral subtraction, SPS 2 'performed on the useful signal of the channel 2 to form together with the useful signal in channel 1, a second reference signal R2.
- the result is a system with 2 reference signals, which are subtracted from the primary signal P.
- the interference is detected in each case with the characteristic of the individual microphones during speech pauses, and a zero point for the speech signal is generated during speech activity.
- FIG. 4 the modification with 2 reference inputs for 'end fire' microphone arrangement or 'broad side' arrangement is used.
- Figure 5 shows the block diagram for the 'end fire' arrangement.
- the beamformer consists of the channel 1 for the speech signal and two reference channels 2, 3. Each reference input is filtered by an adaptive filter 'H 1 ', or 'H 2 '. Filter balancing is performed with a multi-channel LMS algorithm.
- a one-sided spectral subtraction is performed by combining two inputs each in the manner described in order to obtain a reference signal. If, for example, a 'broad side array' with 3 microphones is assumed, this results in 6 combinations for pairing. Taking into account that for each pair the one-sided spectral subtraction is optionally performed on one or the other channel, so doubles the number of combinations and thus the number of Referenzkale.
- An array of multiple microphones uses a limited number of possible combinations.
- the invention is not limited to the recording of the useful signals by microphones, but receiving systems such as antennas can be used.
- Useful signals can be any kind of acoustic and electrical signals.
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- Engineering & Computer Science (AREA)
- Computational Linguistics (AREA)
- Quality & Reliability (AREA)
- Signal Processing (AREA)
- Health & Medical Sciences (AREA)
- Audiology, Speech & Language Pathology (AREA)
- Human Computer Interaction (AREA)
- Physics & Mathematics (AREA)
- Acoustics & Sound (AREA)
- Multimedia (AREA)
- Circuit For Audible Band Transducer (AREA)
- Noise Elimination (AREA)
Claims (11)
- Procédé pour la production d'un signal de référence parasite R pour la réduction du bruit d'un signal utile primaire, qui est formé par la combinaison de signaux d'au moins deux canaux, en particulier de canaux vocaux, où les signaux sont traités ensemble par paire et où à chaque fois un seul des signaux traités par paire est soumis à une soustraction spectrale et est pris en compte pour la formation d'une différence avec l'autre signal, si bien qu'il en résulte au final un signal de référence parasite R, qui ne contient sensiblement plus que le signal parasite lui-même, à titre de référence.
- Procédé selon la revendication 1, caractérisé en ce que le signal utile primaire est mis en circuit à titre de réseau différentiel (DA) de deux canaux (1, 2).
- Procédé selon la revendication 1, caractérisé en ce que le signal utile primaire est mis en circuit à titre de réseau de somme et de différence (SD) de deux canaux (1,2).
- Procédé selon l'une des revendications précédentes, caractérisé en ce que le signal de référence parasite avec l'élargissement supplémentaire de la soustraction spectrale monolatérale est produit sous une forme différentielle de telle sorte que la différence entre le signal utile déparasité provenant d'un canal (1) et le signal utile provenant d'un autre canal (2) soit émis sur un filtre adaptatif (H1) et que le signal de référence parasite (R) filtré soit soustrait ensuite du signal utile primaire (P).
- Procédé selon l'une des revendications 1 à 3, caractérisé en ce qu'une soustraction spectrale (SPS1) est réalisée sur un premier canal (1) pour le signal utile et est émise conjointement avec le signal utile dans un deuxième canal (2) sur un filtre adaptatif (H1) et un premier signal de référence (R1) est formé, en ce qu'une autre soustraction spectrale (SPS2) est réalisée sur le signal utile du deuxième canal (2) et est émise conjointement avec le signal utile provenant du premier canal (1) sur un filtre adaptatif (H2) dans un autre canal (3) et un deuxième signal de référence (R2) est formé, et en ce que les deux signaux de référence (R1, R2) sont soustraits du signal utile primaire (P).
- Procédé selon l'une des revendications précédentes, caractérisé en ce que les filtres (H1, H2) sont adaptés à l'algorithme LMS dans la zone de la durée ou dans la zone de la fréquence.
- Procédé selon l'une des revendications précédentes, caractérisé en ce que le signal utile est enregistré par des microphones.
- Procédé selon l'une des revendications précédentes, caractérisé en ce que l'on utilise comme signal utile un signal vocal.
- Procédé selon l'une des revendications précédentes, caractérisé en ce que la soustraction spectrale est ajustée en continu dans son efficacité avec un paramètre.
- Procédé selon la revendication 9, caractérisé en ce que le paramètre est formé en tant que valeur minimale d'un coefficient de filtre de la soustraction spectrale pour chaque indice de fréquence.
- Procédé selon l'une des revendications précédentes, caractérisé en ce que l'on réalise une soustraction spectrale pour l'émission d'un signal de référence en cas de présence de plus de deux signaux d'entrée, en combinant les deux entrées à chaque fois.
Applications Claiming Priority (2)
Application Number | Priority Date | Filing Date | Title |
---|---|---|---|
DE10118653 | 2001-04-14 | ||
DE10118653A DE10118653C2 (de) | 2001-04-14 | 2001-04-14 | Verfahren zur Geräuschreduktion |
Publications (3)
Publication Number | Publication Date |
---|---|
EP1251493A2 EP1251493A2 (fr) | 2002-10-23 |
EP1251493A3 EP1251493A3 (fr) | 2003-11-19 |
EP1251493B1 true EP1251493B1 (fr) | 2006-08-16 |
Family
ID=7681629
Family Applications (1)
Application Number | Title | Priority Date | Filing Date |
---|---|---|---|
EP02008011A Expired - Lifetime EP1251493B1 (fr) | 2001-04-14 | 2002-04-10 | Procédé pour la réduction du bruit avec fréquence parasite auto-adaptative |
Country Status (5)
Country | Link |
---|---|
US (1) | US7020291B2 (fr) |
EP (1) | EP1251493B1 (fr) |
JP (1) | JP4588966B2 (fr) |
AT (1) | ATE336782T1 (fr) |
DE (2) | DE10118653C2 (fr) |
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DE4307688A1 (de) * | 1993-03-11 | 1994-09-15 | Daimler Benz Ag | Verfahren zur Geräuschreduktion für gestörte Sprachkanäle |
DE4243831A1 (de) * | 1992-12-23 | 1994-06-30 | Daimler Benz Ag | Verfahren zur Laufzeitschätzung an gestörten Sprachkanälen |
US5574824A (en) * | 1994-04-11 | 1996-11-12 | The United States Of America As Represented By The Secretary Of The Air Force | Analysis/synthesis-based microphone array speech enhancer with variable signal distortion |
JP2758846B2 (ja) * | 1995-02-27 | 1998-05-28 | 埼玉日本電気株式会社 | ノイズキャンセラ装置 |
US6717991B1 (en) * | 1998-05-27 | 2004-04-06 | Telefonaktiebolaget Lm Ericsson (Publ) | System and method for dual microphone signal noise reduction using spectral subtraction |
JP4163294B2 (ja) * | 1998-07-31 | 2008-10-08 | 株式会社東芝 | 雑音抑圧処理装置および雑音抑圧処理方法 |
-
2001
- 2001-04-14 DE DE10118653A patent/DE10118653C2/de not_active Expired - Fee Related
-
2002
- 2002-04-10 EP EP02008011A patent/EP1251493B1/fr not_active Expired - Lifetime
- 2002-04-10 AT AT02008011T patent/ATE336782T1/de not_active IP Right Cessation
- 2002-04-10 DE DE50207832T patent/DE50207832D1/de not_active Expired - Lifetime
- 2002-04-11 JP JP2002108951A patent/JP4588966B2/ja not_active Expired - Fee Related
- 2002-04-12 US US10/121,180 patent/US7020291B2/en not_active Expired - Lifetime
Cited By (1)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
CN102077607B (zh) * | 2008-05-02 | 2014-12-10 | Gn奈康有限公司 | 组合至少两个音频信号的方法和包括至少两个麦克风的麦克风系统 |
Also Published As
Publication number | Publication date |
---|---|
JP4588966B2 (ja) | 2010-12-01 |
EP1251493A2 (fr) | 2002-10-23 |
US7020291B2 (en) | 2006-03-28 |
DE50207832D1 (de) | 2006-09-28 |
DE10118653C2 (de) | 2003-03-27 |
ATE336782T1 (de) | 2006-09-15 |
DE10118653A1 (de) | 2002-10-24 |
JP2002374589A (ja) | 2002-12-26 |
EP1251493A3 (fr) | 2003-11-19 |
US20020176589A1 (en) | 2002-11-28 |
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