EP1251493A2 - Method for noise reduction with self-adjusting spurious frequency - Google Patents
Method for noise reduction with self-adjusting spurious frequency Download PDFInfo
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- EP1251493A2 EP1251493A2 EP02008011A EP02008011A EP1251493A2 EP 1251493 A2 EP1251493 A2 EP 1251493A2 EP 02008011 A EP02008011 A EP 02008011A EP 02008011 A EP02008011 A EP 02008011A EP 1251493 A2 EP1251493 A2 EP 1251493A2
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- 238000000034 method Methods 0.000 title claims description 17
- 230000009467 reduction Effects 0.000 title claims description 6
- 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 6
- 238000003491 array Methods 0.000 description 4
- 230000015572 biosynthetic process Effects 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
- 238000013459 approach Methods 0.000 description 1
- 230000002238 attenuated effect Effects 0.000 description 1
- 230000033228 biological regulation Effects 0.000 description 1
- 239000002775 capsule Substances 0.000 description 1
- 238000005314 correlation function Methods 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
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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
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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
- 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 reducing the noise of a disturbed useful signal for example a speech signal, music signal, etc.
- spectral subtraction is the low complexity and that the disturbed useful signal is only required in one variant (only 1 channel).
- Disadvantages are the signal delay (due to the block processing in the spectral range), the limited maximum achievable noise reduction and the difficulty in compensating for unsteady noise.
- Stationary noise can be reduced by 12dB, for example, while the voice quality is still good.
- Microphone arrays used. Of The various microphone arrays are such for many practical applications particularly interesting, those with small geometric dimensions for the microphone arrangement get along. There are small differential microphone arrays (also superdirective Called arrays) and an adaptive variant of this microphone arrangement, the LMS (least mean square) algorithm is used for adaptation.
- LMS least mean square
- two microphones are delay compensated for two Types subtracted in such a way that a virtual microphone with a cardioid polar pattern to the speaker and a 'virtual' microphone with cardioid characteristics from Speakers turned away.
- the runtime compensation corresponds to the time the Sound required for the distance between the two microphones, e.g. 1.5cm. It results a "back-to-back" cardioid polar pattern. That to the speaker
- Directional microphone is the primary signal for the adaptive filter and the opposite directional 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 sample values.
- the combination of two individual microphones with omnidirectional characteristics described above results in a cardioid polar pattern to the speaker and an opposing cardioid polar pattern as an 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 that are attenuated when the cardioid polar pattern is formed.
- the arrangement of the microphones M one behind the other according to FIG. 1 is referred to as an "end fire array", in contrast the arrangement of the microphones next to one another is referred to as a "broad side array”.
- Figure 2 shows an arrangement for a "broad side array" with two microphones at a distance, with the help of spectral subtraction (SPS) the two microphone signals be preprocessed.
- SPS spectral subtraction
- a runtime compensation with the all-pass all between the two channels is executed and serves to compensate for movements of the speaker.
- the sum of the two preprocessed 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 generalized sidelobe canceller '.
- the adaptation is carried out with the LMS algorithm, whereby the LMS is implemented in the frequency domain.
- a postprocessing of the Microphone signals are modified with a cross correlation function in the frequency domain carried out.
- the basic structure with spectral preprocessing using SPS, beam formation and post-processing (post) is in the patent EP 0615226B1 described, wherein an exact specification of the beam former has not been made.
- Figure 3 shows an overview of circuit arrangements of microphones for formation of the directional characteristics for two microphones.
- the two individual microphones themselves can already have a kidney-shaped characteristic or the so-called spherical characteristic.
- ALL denotes an all-pass for the runtime compensation.
- 'Gain' is a Gain compensation between the two channels which is required in practice to equalize the sensitivity of the microphone capsules.
- the direction of response in the polar diagrams of the directional characteristics is 90 °.
- the first 3 arrangements a, b and c are suitable as voice channels, since there is a maximum at 90 ° and there is attenuation 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 an interference reference.
- the zero at 90 ° in the polar diagram is necessary so that no speech components get into the reference channel. Speech components in the reference channel partially compensate for the speech.
- a zero in the direction of the speaker is set for the interference reference. However, this will not be the case in practical applications. The result is that speech components are treated like interference signals and are thus removed from the actual speech signal.
- Beamformers are usually only adapted during the pauses in speech, so that no adaptation takes place To enable language components. Nevertheless, even in this case, in the reference existing speech components are compensated because they are always superimposed on the noise.
- the present invention is therefore based on the object of a method for noise reduction specify with which a crosstalk of the useful signal in the interference reference signal is minimized.
- the invention has the advantage that significantly fewer useful signal components, e.g. speech components are present in the interference reference signal than with previous methods.
- the elimination the disruptive parts of the language is therefore under real conditions with reflections of the Speech signal in real rooms such as possible in the motor vehicle.
- the invention is based on the fact that one-sided formation of the interference reference signal spectral subtraction is performed. It is essential that the spectral subtraction to form a reference signal only takes place on one channel, which with 'one-sided' referred to as. One channel thus contains useful and interference signals, the second channel after spectral subtraction only contains useful signals. Subsequent subtraction The useful part of the two channels is subtracted and the interference remains. This difference is the interference reference signal.
- E.g. Microphones are used to record voice signals, so the Voice signals processed in such a way that the interference reference signal is a zero to the speaker in the form of a kidney-shaped or an eight-shaped characteristic.
- the one-sided spectral subtraction leads to a self-controlling regulation of the Characteristic in such a way that the zero point only arises during speech activity. During breaks in speech The one-sided spectral subtraction means that nothing or only a little Signal is subtracted and thus approximately the characteristics of the individual microphone (e.g. kidney or ball) is available for the disorder.
- the ideal zero for the speech signal in the reference is only with an ideal spectral subtraction achieved in the acoustic free field.
- An ideal spectral subtraction gives the undisturbed speech signal as an output signal and would then continue each Make editing unnecessary.
- the spectral subtraction in practice gives only one 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, the language components of the reference decrease 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 3 block diagrams with one-sided spectral subtraction for the reference input.
- the primary useful signal P of the beam polymer eg voice signal
- FIG. 4a the primary useful signal P of the beam polymer (eg voice signal) is connected as a differential array DA for 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 expansion of the one-sided spectral subtraction in differential form according to the arrangement d and e in FIG. 3.
- the difference between the useful signal in channel 2 and the suppressed useful signal from channel 1 is passed 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 FIG. 5 consists in that the one-sided spectral subtraction, PLC 1 ', is carried out once on channel 1 for the useful signal in order to form a first reference signal R1 together with the useful signal in channel 2.
- the one-sided spectral subtraction, SPS 2 ' is carried out a second time on the useful signal of channel 2 in order to form a second reference signal R2 together with the useful signal in channel 1.
- the result is a system with 2 reference signals that are subtracted from the primary signal P.
- the disturbance is recorded with the characteristics of the individual microphones during the speech pauses, and a zero point for the speech signal is generated during speech activity.
- FIG. 4 shows the modification with two reference inputs for an 'end fire' microphone arrangement or a 'broad side' arrangement.
- Figure 5 shows the block diagram for the 'end fire' arrangement.
- the beamformer consists of 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 '.
- the filter adjustment is carried out using a multi-channel LMS algorithm.
- a combination of 2 inputs in the described manner is used to perform a one-sided spectral subtraction in order to obtain a reference signal. If, for example, a 'broad side array' with 3 microphones is assumed, 6 combinations result for the pair formation. If one takes into account that for each pair the one-sided spectral subtraction is carried out either on one or the other channel, the number of combinations and thus the number of reference channels is doubled. In the case of an array of several microphones, a limited number from the possible combinations is used.
- the invention is not restricted to the recording of the useful signals by microphones, but rather reception systems such as antennas can be used.
- Useful signals can be any type of acoustic and electrical signals.
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Abstract
Description
Die Erfindung betrifft ein Verfahren zur Geräuschreduktion nach dem Oberbegriff des
Patentanspruchs 1.
Ein häufig verwendetes Verfahren zur Geräuschreduktion eines gestörten Nutzsignals,
z.B. ein Sprachsignal, Musiksignal etc. ist die spektrale Subtraktion. Vorteil der spektralen
Subtraktion ist die geringe Komplexität und daß das gestörte Nutzsignal nur in
einer Variante (nur 1 Kanal) benötigt wird. Nachteil ist die Signalverzögerung (bedingt
durch die Blockverarbeitung im Spektralbereich), die begrenzte maximal erreichbare
Geräuschreduktion und die Schwierigkeit instationäre Geräusche zu kompensieren.
Stationäre Geräusche können bei noch guter Sprachqualität z.B. um 12dB reduziert
werden.The invention relates to a method for noise reduction according to the preamble of patent claim 1.
A frequently used method for reducing the noise of a disturbed useful signal, for example a speech signal, music signal, etc., is spectral subtraction. The advantage of spectral subtraction is the low complexity and that the disturbed useful signal is only required in one variant (only 1 channel). Disadvantages are the signal delay (due to the block processing in the spectral range), the limited maximum achievable noise reduction and the difficulty in compensating for unsteady noise. Stationary noise can be reduced by 12dB, for example, while the voice quality is still good.
Wird eine höhere Geräuschreduktion oder eine bessere Sprachqualität verlangt, sind mehrere Aufnahmekanäle erforderlich. Es werden z.B. Mikrofon-Arrays verwendet. Von den verschiedenen Mikrofon-Arrays sind für viele praktische Anwendungen solche besonders interessant, die mit kleinen geometrischen Abmessungen für die Mikrofonanordnung auskommen. Es werden kleine differentielle Mikrofon-Arrays (auch superdirektive Arrays genannt) gebildet und eine adaptive Variante dieser Mikrofonanordnung, wobei zur Adaption der LMS(least mean square )-Algorithmus verwendet wird. Bei der adaptiven Form dieses Arrays werden zwei Mikrofone laufzeitkompensiert auf zwei Arten derart subtrahiert, daß ein virtuelles'Mikrofon mit nierenförmiger Richtcharakteristik zum Sprecher und ein 'virtuelles' Mikrofon mit nierenförmiger Charakteristik vom Sprecher abgewandt entsteht. Die Laufzeitkompensation entspricht der Zeit, die der Schall für die Distanz zwischen den beiden Mikrofonen benötigt, z.B. 1,5cm. Es ergibt sich eine "Rücken-an-Rücken" nierenförmige Richtcharakteristik. Das zum Sprecher gerichtet Mikrofon ist das primäre Signal für das adaptive Filter und das entgegengesetzt gerichtete Mikrofon ist das Referenzsignal der Störung.If a higher noise reduction or a better voice quality is required, multiple recording channels required. E.g. Microphone arrays used. Of The various microphone arrays are such for many practical applications particularly interesting, those with small geometric dimensions for the microphone arrangement get along. There are small differential microphone arrays (also superdirective Called arrays) and an adaptive variant of this microphone arrangement, the LMS (least mean square) algorithm is used for adaptation. at In the adaptive form of this array, two microphones are delay compensated for two Types subtracted in such a way that a virtual microphone with a cardioid polar pattern to the speaker and a 'virtual' microphone with cardioid characteristics from Speakers turned away. The runtime compensation corresponds to the time the Sound required for the distance between the two microphones, e.g. 1.5cm. It results a "back-to-back" cardioid polar pattern. That to the speaker Directional microphone is the primary signal for the adaptive filter and the opposite directional microphone is the reference signal of the interference.
Figure 1 zeigt eine adaptive Anordnung für einen Strahlformer. Der Laufzeitausgleich
mit einem Allpaß ALL wird durch Verschiebung um ganze Abtastwerte realisiert. Durch
die oben beschriebene Kombination zweier Einzelmikrofone mit Kugelcharakteristik
ergibt sich eine nierenförmige Richtcharakteristik zum Sprecher und eine entgegengesetzt
gerichtete nierenförmige Richtcharakteristik als Störreferenz. Das adaptive
Filter H 1 wird im Zeitbereich mit dem LMS (least mean square)-Algorithmus adaptiert.
Ein Tiefpaß TP am Systemausgang hebt tiefe Frequenzanteile an, die bei der Bildung der
nierenförmigen Richtcharakteristik gedämpft werden.
Die Anordnung der Mikrofone M hintereinander gemäß Figur 1 wird als ,end fire array'
bezeichnet, im Gegensatz wird die Anordnung der Mikrofone nebeneinander mit ,broad
side array' bezeichnet.Figure 1 shows an adaptive arrangement for a beamformer. The runtime compensation with an all-pass ALL is realized by shifting by whole sample values. The combination of two individual microphones with omnidirectional characteristics described above results in a cardioid polar pattern to the speaker and an opposing cardioid polar pattern as an 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 that are attenuated when the cardioid polar pattern is formed.
The arrangement of the microphones M one behind the other according to FIG. 1 is referred to as an "end fire array", in contrast the arrangement of the microphones next to one another is referred to as a "broad side array".
Figure 2 zeigt eine Anordnung für ein "broad side array" aus zwei Mikrofonen im Abstand, wobei mit Hilfe der spektralen Subtraktion (SPS) die beiden Mikrofonsignale vorverarbeitet werden. Ein Laufzeitausgleich mit dem Allpaß All zwischen beiden Kanälen wird ausgeführt und dient dem Ausgleich von Bewegungen des Sprechers. Die Summe der beiden vorverarbeiteten Mikrofonsignale bildet den primären Eingang und die Differenz den Referenzeingang für ein adaptives Filter H 1. Das adaptive Filter in dieser Anordnung mit Summen- und Differenzeingang wird auch als ,generalized sidelobe canceller' bezeichnet. Die Adaption erfolgt mit dem LMS-Algorithmus, wobei die Implementierung des LMS im Frequenzbereich erfolgt. Eine Nachverarbeitung der Mikrofonsignale wird mit einer modifizierten Kreuzkorrelationsfunktion im Frequenzbereich durchgeführt. Die grundlegende Struktur mit spektraler Vorverarbeitung mittels SPS, Strahlfomung und Nachverarbeitung (Post) ist in der Patentschrift EP 0615226B1 beschrieben, wobei eine genaue Spezifizierung des Strahlformers nicht erfolgt ist.Figure 2 shows an arrangement for a "broad side array" with two microphones at a distance, with the help of spectral subtraction (SPS) the two microphone signals be preprocessed. A runtime compensation with the all-pass all between the two channels is executed and serves to compensate for movements of the speaker. The The sum of the two preprocessed 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 generalized sidelobe canceller '. The adaptation is carried out with the LMS algorithm, whereby the LMS is implemented in the frequency domain. A postprocessing of the Microphone signals are modified with a cross correlation function in the frequency domain carried out. The basic structure with spectral preprocessing using SPS, beam formation and post-processing (post) is in the patent EP 0615226B1 described, wherein an exact specification of the beam former has not been made.
Figure 3 zeigt einen Überblick über Schaltungsanordnungen von Mikrofonen zur Bildung der Richtcharakteristiken für zwei Mikrofone. Die beiden einzelnen Mikrofone selbst können bereits eine nierenförmige Charakteristik haben oder die sogenannte Kugelcharakteristik. "ALL" bezeichnet einen Allpaß für den Laufzeitausgleich. ,Gain' ist ein Verstärkungsausgleich zwischen beiden Kanälen der in der Praxis erforderlich ist, um die Empfindlichkeit der Mikrofonkapseln anzugleichen.Figure 3 shows an overview of circuit arrangements of microphones for formation of the directional characteristics for two microphones. The two individual microphones themselves can already have a kidney-shaped characteristic or the so-called spherical characteristic. "ALL" denotes an all-pass for the runtime compensation. 'Gain' is a Gain compensation between the two channels which is required in practice to equalize the sensitivity of the microphone capsules.
Die Einsprechrichtung in den Polardiagrammen der Richtcharakteristiken ist 90°. Die
ersten 3 Anordnungen a, b und c sind als Sprachkanal geeignet, da bei 90° ein Maximum
vorliegt und für die weiteren Richtungen eine Dämpfung vorhanden ist. Anordnung
a und b führen auf die gleiche Richtcharakteristik. Die Anordnungen a, b werden als
Summen- oder Differenz Array und Anordnung c als differentielles Array bezeichnet.
Die Anordnungen d und e haben eine Nullstelle bei 90°im Polardiagramm und sind
damit als Störreferenz geeignet. Die Nullstelle bei 90° im Polardiagramm ist notwendig,
damit keine Sprachanteile in den Referenzkanal gelangen. Sprachanteile im Referenzkanal
führen zur teilweisen Kompensation der Sprache.
Unter idealen Bedingungen wird sich gemäß Anordnung d und e für die Störreferenz
eine Nullstelle in Richtung zum Sprecher einstellen. In praktischen Anwendungen wird
dies jedoch nicht der Fall sein. Die Folge ist, daß Sprachanteile wie Störsignale behandelt
werden und damit vom eigentlichen Sprachsignal entfernt werden.The direction of response in the polar diagrams of the directional characteristics is 90 °. The first 3 arrangements a, b and c are suitable as voice channels, since there is a maximum at 90 ° and there is attenuation 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 an interference reference. The zero at 90 ° in the polar diagram is necessary so that no speech components get into the reference channel. Speech components in the reference channel partially compensate for the speech.
Under ideal conditions, according to the arrangement d and e, a zero in the direction of the speaker is set for the interference reference. However, this will not be the case in practical applications. The result is that speech components are treated like interference signals and are thus removed from the actual speech signal.
Strahlformer werden meist nur in den Sprachpausen adaptiert, um keine Adaption an Sprachanteile zu ermöglichen. Dennoch werden auch in diesem Fall in der Referenz vorhandene Sprachanteile kompensiert, da sie dem Geräusch stets überlagert sind.Beamformers are usually only adapted during the pauses in speech, so that no adaptation takes place To enable language components. Nevertheless, even in this case, in the reference existing speech components are compensated because they are always superimposed on the noise.
Eine andere Vorgehensweise ist die Verstärkung von Kanälen anzugleichen, damit bei deren Subtraktion im Idealfall eine Nullstelle erzeugt wird. Dies ist notwendig, da Mikrofone aus der Serienfertigung Toleranzen ausweisen. In den Anordnungen der Figur 3 ist dies mit dem Funktionsblock ,Gain' berücksichtigt, der unterschiedliche Mikrofon-Empfindlichkeiten ausgleicht.Another approach is to equalize the gain of channels so that whose subtraction ideally creates a zero. This is necessary because Series production microphones show tolerances. In the arrangements of the figure 3 this is taken into account with the function block 'Gain', which has different microphone sensitivities balances.
In Anwendungen wird trotz Empfindlichkeitsausgleich mit 'Gain' dennoch keine Nullstelle für das Sprachsignal in der Referenz eingestellt. Nur unter der Voraussetzung, daß das Mikrofon im akustischen Freifeld betrieben wird (ohne Reflexionen), können die Sprachanteile vollständig kompensiert werden. Reale Anwendungen haben bedingt durch Reflexionen einen gewissen Schallanteil aus unterschiedlichen Richtungen, der eine Nullstelle für das Sprachsignal nicht entstehen läßt. Es wird sich bei Anordnungen gemäß Figur 1 oder Figur 2 stets ein gewisser Sprachanteil in dem Referenzsignal des Strahlformers wiederfinden, der zu Sprachverzerrungen führt.Despite sensitivity compensation with 'Gain', there is still no zero in applications set for the speech signal in the reference. Provided that the microphone is operated in the acoustic free field (without reflections) Parts of the language are fully compensated. Real applications have conditional by reflections a certain proportion of sound from different directions, the a zero for the voice signal does not arise. There will be orders According to FIG. 1 or FIG. 2, there is always a certain amount of speech in the reference signal of the Find beamformers, which leads to speech distortions.
Der vorliegende Erfindung liegt deshalb die Aufgabe zugrunde, ein Verfahren zur Geräuschreduktion anzugeben, mit dem ein Übersprechen des Nutzsignals in das Störreferenzsignal minmiert wird.The present invention is therefore based on the object of a method for noise reduction specify with which a crosstalk of the useful signal in the interference reference signal is minimized.
Die Erfindung ist in Anspruch 1 angegeben. Vorteilhafte Ausgestaltungen und Weiterbildungen sind den Unteransprüchen zu entnehmen.The invention is set out in claim 1. Advantageous refinements and developments can be found in the subclaims.
Die Erfindung hat den Vorteil, daß deutlich weniger Nutzsignalanteile, z.B. Sprachanteile im Störreferenzsignal vorhanden sind als mit den bisherigen Verfahren. Die Beseitigung der störenden Sprachanteile ist damit unter realen Bedingungen mit Reflexionen des Sprachsignals in realen Räumen wie z.B. im Kraftfahrzeug möglich.The invention has the advantage that significantly fewer useful signal components, e.g. speech components are present in the interference reference signal than with previous methods. The elimination the disruptive parts of the language is therefore under real conditions with reflections of the Speech signal in real rooms such as possible in the motor vehicle.
Die Erfindung geht davon aus, daß zur Bildung des Störreferenzsignals eine einseitige spektrale Subtraktion durchgeführt wird. Wesentlich ist, daß die spektrale Subtraktion zur Bildung eines Referenzsignals nur an einem Kanal stattfindet, was mit 'einseitig' bezeichnet wird. Der eine Kanal enthält damit Nutz- und Störsignale, der zweite Kanal nach der spektralen Subtraktion enthält nur Nutzsignale. Bei der anschließenden Subtraktion der beiden Kanäle wird der Nutzanteil subtrahiert und es verbleibt die Störung. Diese Differenz ist das Störreferenzsignal. The invention is based on the fact that one-sided formation of the interference reference signal spectral subtraction is performed. It is essential that the spectral subtraction to form a reference signal only takes place on one channel, which with 'one-sided' referred to as. One channel thus contains useful and interference signals, the second channel after spectral subtraction only contains useful signals. Subsequent subtraction The useful part of the two channels is subtracted and the interference remains. This difference is the interference reference signal.
Werden z.B. Mikrofone zur Aufnahme von Sprachsignalen verwendet, so werden die Sprachsignale derart verarbeitet, daß das Störreferenzsignal eine Nullstelle zum Sprecher in der Form einer nierenförmigen oder einer achtförmigen Charakteristik aufweist. Die einseitige spektrale Subtraktion führt zu einer selbststeuernden Regelung der Charakteristik, derart, daß die Nullstelle nur bei Sprachaktivität entsteht. In Sprachpausen führt die einseitige spektrale Subtraktion dazu, daß nichts oder nur ein geringes Signal subtrahiert wird und damit näherungsweise die Charakteristik des Einzelmikrofons (z.B. Niere oder Kugel) für die Störung zur Verfügung steht.E.g. Microphones are used to record voice signals, so the Voice signals processed in such a way that the interference reference signal is a zero to the speaker in the form of a kidney-shaped or an eight-shaped characteristic. The one-sided spectral subtraction leads to a self-controlling regulation of the Characteristic in such a way that the zero point only arises during speech activity. During breaks in speech The one-sided spectral subtraction means that nothing or only a little Signal is subtracted and thus approximately the characteristics of the individual microphone (e.g. kidney or ball) is available for the disorder.
Die ideale Nullstelle für das Sprachsignal in der Referenz wird nur mit einer idealen spektralen Subtraktion im akustischen Freifeld erreicht. Eine ideale spektrale Subtraktion ergibt das ungestörte Sprachsignal als Ausgangssignal und würde dann jede weiter Bearbeitung unnötig machen. Die spektrale Subtraktion in der Praxis ergibt nur eine gute Annäherung des Sprachsignals mit Geräuschresten in den Sprachpausen. Da die einseitige spektrale Subtraktion ergänzend zu der Mikrofon-Nullstelle eingesetzt wird, vermindern sich die Sprachanteile der Referenz deutlich.The ideal zero for the speech signal in the reference is only with an ideal spectral subtraction achieved in the acoustic free field. An ideal spectral subtraction gives the undisturbed speech signal as an output signal and would then continue each Make editing unnecessary. The spectral subtraction in practice gives only one 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, the language components of the reference decrease significantly.
Das Restgeräusch der spektralen Subtraktion in Sprachpausen wird mit einem Parameter
eingestellt, dem ,spectral floor'. Der spectral floor b ist der minimale Wert eines
Filterkoeffizienten W der spektralen Subtraktion bei jedem Frequenzindex i. Das Ausgangssignal
Y(i) ergibt sich durch Multiplikation der Filterkoeffizienten W(i) mit dem
Eingangswert X(i):
Der maximale Wert für W ist 1 (Ausgang =Eingang). Wird b=1 gewählt, ist die spektrale Subtraktion praktisch ausgeschaltet.. Mit b=0 erreicht die spektrale Subtraktion die maximale Wirksamkeit. In der Praxis ergibt sich mit b=0 eine schlechte Sprachqualität. The maximum value for W is 1 (output = input). If b = 1 is chosen, the spectral is Subtraction practically switched off. With b = 0 the spectral subtraction reaches the maximum effectiveness. In practice, b = 0 results in poor speech quality.
Mit dem Parameter b ergibt sich für die vorliegende Erfindung die Möglichkeit die einseitige spektrale Subtraktion in ihrer Wirksamkeit kontinuierlich einzustellen. Mit einem Wert von z.B. b=0.25 wird eine Geräuschunterdrückung von ca. 12dB und eine gute Sprachqualität erzielt.With parameter b, the possibility arises for the present invention to be one-sided adjust the effectiveness of spectral subtraction continuously. With a Value of e.g. b = 0.25 will have a noise suppression of about 12dB and a good one Voice quality achieved.
Die Erfindung wird im folgenden anhand von Ausführungsbeispielen unter Bezugnahme auf schematische Zeichnungen näher erläutert.The invention is described below with reference to exemplary embodiments explained in more detail on schematic drawings.
Figur 4 zeigt 3 Blockschaltbilder mit einseitiger spektraler Subtraktion für den Referenzeingang.
In Figur 4a ist das primäre Nutzsignal P des Strahlfomers (z.B. Sprachsignal)
als differentielles Array DA für die Kanäle 1, 2 geschaltet ist (Anordnung c in
Figur 3). Figur 4b, 4c zeigt eine Schaltung des Primärsignals P als Summen- und Differenz
Array SD (Anordnung a und b in Figur 3).
Der Störreferenzeingang verarbeitet das Referenzsignal R mit der zusätzlichen Erweiterung
der einseitigen spektralen Subtraktion in differentieller Form gemäß den Anordnung
d und e in Figur 3. Die Differenz aus Nutzsignal in Kanal 2 und entstörtem
Nutzsignal aus Kanal 1 wird auf das adaptive Filter H 1 gegeben. Das adaptive Filter H1
wird im Zeitbereich oder in einer äquivalenten Form im Frequenzbereich mit dem LMS -
Algorithmus adaptiert. Das gefilterte Störreferenzsignal R wird anschließend vom
primären Nutzsignal P subtrahiert.FIG. 4 shows 3 block diagrams with one-sided spectral subtraction for the reference input. In FIG. 4a, the primary useful signal P of the beam polymer (eg voice signal) is connected as a differential array DA for 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 expansion of the one-sided spectral subtraction in differential form according to the arrangement d and e in FIG. 3. The difference between the useful signal in
Eine weitere Ausgestaltung der Erfindung gemäß Figur 5 besteht darin, daß die einseitige
spektrale Subtraktion ,SPS1'einmal am Kanal 1 für das Nutzsignal durchgeführt
wird, um damit zusammen mit dem Nutzsignal in Kanal 2 einen erstes Referenzsignal
R1 zu bilden. Ein zweites Mal wird die einseitige spektrale Subtraktion ,SPS2'am Nutzsignal
des Kanal 2 durchgeführt, um zusammen mit dem Nutzsignal in Kanal 1 ein
zweites Referenzsignal R2 zu bilden. Es entsteht ein System mit 2 Referenzsignalen, die
vom Primärsignal P subtrahiert werden. Bei Sprachsignalen wird in den Sprachpausen
die Störung jeweils mit der Charakteristik der Einzelmikrofone erfaßt und bei Sprachaktivität
eine Nullstelle für das Sprachsignal erzeugt. A further embodiment of the invention according to FIG. 5 consists in that the one-sided spectral subtraction, PLC 1 ', is carried out once on channel 1 for the useful signal in order to form a first reference signal R1 together with the useful signal in
Entsprechend den Erläuterungen zu den Blockschaltbildern der Figur 4 wird die Abwandlung
mit 2 Referenzeingängen für 'end fire' Mikrofonanordnung oder ,broad side'
Anordnung verwendet. Figure 5 zeigt das Blockschaltschild für die ,end fire' Anordnung.
Der Strahlformer besteht aus dem Kanal 1 für das Sprachsignal und zwei Referenzkanälen
2, 3. Jeder Referenzeingang wird von einem adaptiven Filter 'H1', bzw. 'H2' gefiltert.
Der Filterabgleich erfolgt mit einem mehrkanaligen LMS-Algorithmus.In accordance with the explanations for the block diagrams in FIG. 4, the modification with two reference inputs is used for an 'end fire' microphone arrangement or a 'broad side' arrangement. Figure 5 shows the block diagram for the 'end fire' arrangement. The beamformer consists of channel 1 for the speech signal and two
Stehen mehr als 2 Eingangssignale zur Verfügung, so wird durch Kombination von jeweils
2 Eingängen in der beschriebenen Weise eine einseitige spektrale Subtraktion
durchgeführt, um ein Referenzsignal zu erhalten. Wird z.B. ein ,broad side array' mit 3
Mikrofonen angenommen, ergeben sich für die Paarbildung 6 Kombinationen. Wird
berücksichtigt, daß bei jedem Paar die einseitige spektrale Subtraktion wahlweise bei
dem einen oder dem anderen Kanal durchgeführt wird, so verdoppelt sich die Anzahl
der Kombinationen und somit die Anzahl der Referenzkänale. Bei einem Array aus
mehreren Mikrofonen wird eine eingeschränkte Anzahl aus den möglichen Kombinationen
verwendet.
Die Erfindung ist nicht auf die Aufzeichnung der Nutzsignale durch Mikrofone beschränkt,
sondern es können Empfangssysteme wie z.B. Antennen verwendet werden.
Nutzsignale können jegliche Art von akustischen und elektrischen Signalen sein.If more than 2 input signals are available, a combination of 2 inputs in the described manner is used to perform a one-sided spectral subtraction in order to obtain a reference signal. If, for example, a 'broad side array' with 3 microphones is assumed, 6 combinations result for the pair formation. If one takes into account that for each pair the one-sided spectral subtraction is carried out either on one or the other channel, the number of combinations and thus the number of reference channels is doubled. In the case of an array of several microphones, a limited number from the possible combinations is used.
The invention is not restricted to the recording of the useful signals by microphones, but rather reception systems such as antennas can be used. Useful signals can be any type of acoustic and electrical signals.
Claims (11)
wobei die Signale paarweise miteinander verarbeitet werden, und
wobei jeweils nur eines der paarweise verarbeiteten Signale einer spektralen Subtraktion unterzogen und zur Differenzbildung mit dem anderen Signal herangezogen wird,Method for generating an interference reference signal R for noise reduction of a primary useful signal, which is formed by combining the signals of at least two channels, in particular voice channels,
the signals are processed in pairs, and
only one of the signals processed in pairs is subjected to spectral subtraction and used to form the difference with the other signal,
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DE10118653A DE10118653C2 (en) | 2001-04-14 | 2001-04-14 | Method for noise reduction |
DE10118653 | 2001-04-14 |
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EP1251493A2 true EP1251493A2 (en) | 2002-10-23 |
EP1251493A3 EP1251493A3 (en) | 2003-11-19 |
EP1251493B1 EP1251493B1 (en) | 2006-08-16 |
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EP02008011A Expired - Lifetime EP1251493B1 (en) | 2001-04-14 | 2002-04-10 | Method for noise reduction with self-adjusting spurious frequency |
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US (1) | US7020291B2 (en) |
EP (1) | EP1251493B1 (en) |
JP (1) | JP4588966B2 (en) |
AT (1) | ATE336782T1 (en) |
DE (2) | DE10118653C2 (en) |
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Also Published As
Publication number | Publication date |
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ATE336782T1 (en) | 2006-09-15 |
DE10118653C2 (en) | 2003-03-27 |
DE50207832D1 (en) | 2006-09-28 |
EP1251493B1 (en) | 2006-08-16 |
JP4588966B2 (en) | 2010-12-01 |
EP1251493A3 (en) | 2003-11-19 |
JP2002374589A (en) | 2002-12-26 |
US7020291B2 (en) | 2006-03-28 |
US20020176589A1 (en) | 2002-11-28 |
DE10118653A1 (en) | 2002-10-24 |
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