EP2293594A1 - Verfahren zur Filterung von seitlichem nichtstationärem Rauschen für ein Multimikrofon-Audiogerät - Google Patents
Verfahren zur Filterung von seitlichem nichtstationärem Rauschen für ein Multimikrofon-Audiogerät Download PDFInfo
- Publication number
- EP2293594A1 EP2293594A1 EP10166119A EP10166119A EP2293594A1 EP 2293594 A1 EP2293594 A1 EP 2293594A1 EP 10166119 A EP10166119 A EP 10166119A EP 10166119 A EP10166119 A EP 10166119A EP 2293594 A1 EP2293594 A1 EP 2293594A1
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- EP
- European Patent Office
- Prior art keywords
- transients
- probability
- speech
- signal
- noise
- 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.)
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Classifications
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
- H04R3/00—Circuits for transducers, loudspeakers or microphones
- H04R3/005—Circuits for transducers, loudspeakers or microphones for combining the signals of two or more microphones
-
- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; SPEECH OR AUDIO CODING OR DECODING
- G10L21/00—Processing of the speech or voice signal 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
- G10L2021/02087—Noise filtering the noise being separate speech, e.g. cocktail party
-
- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; SPEECH OR AUDIO CODING OR DECODING
- G10L21/00—Processing of the speech or voice signal 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
- G10L2021/02166—Microphone arrays; Beamforming
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- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; SPEECH OR AUDIO CODING OR DECODING
- G10L21/00—Processing of the speech or voice signal 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
- H04R2201/00—Details of transducers, loudspeakers or microphones covered by H04R1/00 but not provided for in any of its subgroups
- H04R2201/10—Details of earpieces, attachments therefor, earphones or monophonic headphones covered by H04R1/10 but not provided for in any of its subgroups
- H04R2201/107—Monophonic and stereophonic headphones with microphone for two-way hands free communication
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
- H04R2430/00—Signal processing covered by H04R, not provided for in its groups
- H04R2430/03—Synergistic effects of band splitting and sub-band processing
Definitions
- the invention relates to the treatment of speech in a noisy environment.
- microphone microphone
- noise that is a disruptive element that can go, in some cases, to make incomprehensible the speaker's words. It is the same if one wants to implement speech recognition techniques, because it is very difficult to perform a form recognition on words embedded in a high noise level.
- Some of these devices provide for the use of several microphones, usually two microphones, and use the average of the signals picked up, or other more complex operations, to obtain a signal with a lower level of interference.
- a so-called beamforming technique makes it possible to create, by software means, a directivity that improves the signal-to-noise ratio, but the performances of this technique are very limited when only two microphones are used.
- conventional techniques are especially adapted to the filtering of diffuse noise, stationary, coming from the surroundings of the device and found at comparable levels in the signals picked up by the two microphones.
- One of the aims of the invention is to take advantage of the multi-microphone structure of the device to operate a spatial detection of these nonstationary noises, then to discriminate, among all the nonstationary components (hereinafter "transients") those which are nonstationary noise components from those which are speech components, and finally to process the captured signal to effectively denoise it while minimizing the distortions introduced by this processing.
- transients those which are nonstationary noise components from those which are speech components
- lateral noise a directional non-stationary noise whose direction of arrival is far from that of the useful signal
- privileged cone the direction or angular sector of the space where the source is located of useful signal (the speech of the speaker) compared to the network of microphones.
- the starting point of the invention consists in associating the properties of temporal and frequency non-stationarity, on the one hand, and spatial directivity, on the other hand, to detect a type of noise that is usually difficult. to discriminate from the speech, then to deduce a probability of presence of the speech which will serve to attenuate this noise.
- the invention relates to a method of denoising a noisy acoustic signal picked up by a plurality of microphones of a multi-microphone audio device operating in a noisy environment.
- the noisy acoustic signal includes a speech component derived from a directional speech source and a noise noise component, said noise component itself including a directional non-stationary side noise component.
- the Figure 1 is a block diagram showing the different modules and functions implemented by the method of the invention as well as their interactions.
- the method of the invention is implemented by software means, which can be broken down and schematized by a number of modules 10 to 24 illustrated Figure 1 .
- the signal which one wishes to denoise comes from a plurality of signals picked up by a network of microphones (which, in the minimum configuration, can be simply a network of two microphones) arranged in a predetermined configuration.
- the microphone array captures the signal transmitted by the useful signal source (speech signal), and the difference in position between the microphones induces a set of phase shifts and amplitude variations in the recording of the signals emitted by the signal source. useful.
- n is the amplitude attenuation due to the energy loss between the position of the sound source s and the microphone
- ⁇ n is the phase shift between the signal transmitted and received by the microphone
- ⁇ n represents the value of the diffuse noise field at the microphone position.
- the delays ⁇ n can then be calculated from the angle ⁇ S , defined as the angle between the mediators of the pairs of microphones (n, m) and the reference direction corresponding to the source s of useful signal.
- the angle ⁇ S is zero.
- the signals X n ( k, l ) can be combined with each other by a simple beamforming pre- filtering technique of the Delay and Sum type which is applied to obtain a partially denoised combined signal X ( k, I ):
- X k ⁇ l 1 NOT ⁇ not ⁇ 1 ⁇ NOT d not k ⁇ .
- this treatment provides only a slight improvement in the signal / noise ratio, of the order of 1 dB only.
- the angle ⁇ S is zero and it is a simple average that is made on both microphones.
- the purpose of this step is to calculate an estimate of the pseudo-stationary noise component V ( k, l ) present on the signal X ( k, l ).
- V ( k, l ) notably an algorithm for estimating the energy of the pseudo-stationary minimum recursive averaging noise component (MCRA) as described by I. Cohen and B. Berdugo, Noise Estimation by Minima Controlled Recursive Averaging for Robust Speech Enhancement, IEEE Signal Processing Letters, Vol. 9, No. 1, pp. 12-15, Jan. 2002 .
- MCRA pseudo-stationary minimum recursive averaging noise component
- Transients refers to all non-stationary signals, including both useful speech and sporadic non-stationary noises, which may have energy equivalent to or sometimes greater than useful speech (passing a vehicle, siren, horn, other people's words etc.).
- the processing performed by the block 16 consists only of calculating a transient probability p ( k, l ) of the presence of transient signals, without distinction between useful speech and non-stationary noise noises.
- the algorithm is as follows:
- TSR min and TSR max are chosen so as to correspond to typical situations, close to reality.
- This calculation takes advantage of the fact that, unlike the pseudo-stationary component of the noise that is diffuse, the transients are often directional, that is to say from a point sound source (such as the mouth of the speaker for useful speech, or the engine of a motorcycle for a lateral noise). It is therefore advisable to calculate the direction of arrival of these signals, which will generally be well defined, and to compare this direction of arrival at the angle ⁇ S corresponding to the direction of useful speech origin), so as to determine whether the non-stationary signal considered is useful or parasitic, and thus to discriminate between useful speech and non-stationary noise.
- the first step is to estimate the direction of arrival of the transient.
- the method used here is based on the use of the probability of transient ( k, 1 ) transient p- transients determined by block 18 as discussed above.
- M 19
- N the number of microphones
- Each angle ⁇ i is tested to determine the one that is closest to the direction of arrival of the non-stationary signal studied. To do this, we consider each pair of microphones ( n, m ) and we calculate an estimator corresponding direction of arrival P n, m ( ⁇ i , k , l ), whose module will be maximum when the angle ⁇ i tested is closest to the direction of arrival of the transient.
- Another method, used here in a preferential way, consists in weighting the estimator P n, m ( ⁇ i , k, l ) by the probability of presence of transients p Transient ( k, l ), and defining a new decision strategy.
- This first rule makes it possible to ensure that on the part ( k, l ) of the signal considered, the probability of presence of a transient and the level of inter-correlation are high enough for the estimate to be consistent.
- This second rule analyzes the content of the "privileged cone", corresponding to the angular sector on which the source s is centered and which has an angular extent of ⁇ 0 .
- This privileged cone is defined by the angles ⁇ such that
- the "lateral noise” will correspond to a signal whose direction of arrival is external to the preferred cone, and it will therefore be considered that a lateral noise is present if
- P New ( ⁇ max , k, l ) is confronted with the values of P New ( ⁇ i , k, l) obtained for other angles, especially those belonging to the privileged cone.
- the rule thus makes it possible to ensure that there is no local maximum in the privileged cone.
- this third rule takes into account the previous frames to avoid false triggers. It only applies to the first frame of a presumed lateral noise, and verifies that P New ( ⁇ max , k, l ) increases significantly with respect to the corresponding data obtained on the five previous frames.
- the parameters ⁇ 1 and ⁇ 2 are chosen so as to correspond to typical situations, close to reality.
- the last two rules are intended to prevent cuts in the detection of a side noise. After a period of detection, they continue to maintain this state for a period of time said hangover , even if the previous decision rules would no longer be verified. This makes it possible to detect possible low energy periods of non-stationary noise.
- the estimate of angle ⁇ max is not made on each frequency band, but on each packet K j of frequency bands.
- the following step which is characteristic of the method of the invention, consists in calculating a probability of presence of speech based on the arrival direction estimation ⁇ ( k ', /) obtained in the manner indicated above.
- the probability p spa ( k, l ) can be calculated in different ways, giving a binary value or multiple values. Two examples of computation p spa ( k, l ) are given below, given that other laws can be used to express p spa ( k, l ) from ⁇ ( k, /).
- the probability p spa ( k, l ) of the presence of speech calculated in block 20, itself dependent on the probability p Transient ( k, l ) of the presence of transients computed at block 16, will be used as input parameter in a classic technique of denoising.
- LSA Log-Spectral Amplitude
- the "OM-LSA” Optimally-Modified Log-Spectral Amplitude ) algorithm improves the calculation of the LSA gain to be applied by weighting it by the conditional probability of presence of speech.
- the probability of presence of speech occurs at two important moments, for the estimation of the energy of the noise and for the calculation of the final gain, and the probability p spa ( k, /) will be used at these two levels. .
- the probability p spa ( k, l ) modulates the forgetting factor in the noise estimate, which is updated more rapidly on the noisy signal X ( k, l ) when the probability of speech is weak, this mechanism completely conditioning the quality of ⁇ Noise ( k, l ).
- G H 1 ( k, l ) being a denoising gain (whose calculation depends on the noise estimate ⁇ Noise ) described in the aforementioned article by Cohen, and G min being a constant corresponding to the denoising applied when speech is considered absent.
- the probability p spa ( k, l ) plays a large role in the determination of G OM-LSA gain ( k, l ).
- the gain is equal to G min and a maximum noise reduction is applied: if, for example, a value of 20 dB is chosen for G min , the non-stationary noises previously detected are attenuated by 20 dB.
- This hybrid probability makes it possible to benefit from the identification of non-stationary noise associated with small values of p spa ( k, l ), and to complete the estimation of the probability p hybrid ( k, l ) on parts ( k, l ). where the direction of arrival estimate ⁇ ( k, l ) has not been defined (producing a probability p spa ( k, l ) forced to the value 1 for safety).
- the hybrid p hybrid probability ( k, l ) thus integrates both the non-stationary noises detected by p spa ( k, 1 ) and the other noises (for example pseudo-stationary) detected by p ( k, l ).
- the last step consists in applying to the signal ⁇ ( k, l ) a fast inverse Fourier transform iFFT to obtain in the time domain the denoised speech signal ⁇ ( t ) .
Applications Claiming Priority (1)
Application Number | Priority Date | Filing Date | Title |
---|---|---|---|
FR0955133A FR2948484B1 (fr) | 2009-07-23 | 2009-07-23 | Procede de filtrage des bruits lateraux non-stationnaires pour un dispositif audio multi-microphone, notamment un dispositif telephonique "mains libres" pour vehicule automobile |
Publications (2)
Publication Number | Publication Date |
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EP2293594A1 true EP2293594A1 (de) | 2011-03-09 |
EP2293594B1 EP2293594B1 (de) | 2011-11-02 |
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Family Applications (1)
Application Number | Title | Priority Date | Filing Date |
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EP10166119A Active EP2293594B1 (de) | 2009-07-23 | 2010-06-16 | Verfahren zur Filterung von seitlichem nichtstationärem Rauschen für ein Multimikrofon-Audiogerät |
Country Status (5)
Country | Link |
---|---|
US (1) | US8370140B2 (de) |
EP (1) | EP2293594B1 (de) |
AT (1) | ATE532345T1 (de) |
ES (1) | ES2377056T3 (de) |
FR (1) | FR2948484B1 (de) |
Cited By (1)
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EP2538409A1 (de) | 2011-06-20 | 2012-12-26 | Parrot | Verfahren zur Geräuschdämpfung für Audio-Gerät mit mehreren Mikrofonen, insbesondere für eine telefonische Freisprechanlage |
Also Published As
Publication number | Publication date |
---|---|
FR2948484B1 (fr) | 2011-07-29 |
US8370140B2 (en) | 2013-02-05 |
ATE532345T1 (de) | 2011-11-15 |
EP2293594B1 (de) | 2011-11-02 |
FR2948484A1 (fr) | 2011-01-28 |
ES2377056T3 (es) | 2012-03-22 |
US20110054891A1 (en) | 2011-03-03 |
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