EP1830349A1 - Verfahren zur Geräuschdämpfung eines Audiosignals - Google Patents
Verfahren zur Geräuschdämpfung eines Audiosignals Download PDFInfo
- Publication number
- EP1830349A1 EP1830349A1 EP07290219A EP07290219A EP1830349A1 EP 1830349 A1 EP1830349 A1 EP 1830349A1 EP 07290219 A EP07290219 A EP 07290219A EP 07290219 A EP07290219 A EP 07290219A EP 1830349 A1 EP1830349 A1 EP 1830349A1
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- speech
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- 238000000034 method Methods 0.000 title claims abstract description 37
- 230000005236 sound signal Effects 0.000 title claims abstract description 8
- 230000009467 reduction Effects 0.000 title abstract description 5
- 238000004422 calculation algorithm Methods 0.000 claims abstract description 41
- 238000001228 spectrum Methods 0.000 claims abstract description 17
- 230000003595 spectral effect Effects 0.000 claims abstract description 11
- 230000001052 transient effect Effects 0.000 claims description 13
- 238000012545 processing Methods 0.000 claims description 10
- 230000003044 adaptive effect Effects 0.000 claims description 8
- 238000004458 analytical method Methods 0.000 claims description 5
- 230000002123 temporal effect Effects 0.000 claims description 5
- 230000010363 phase shift Effects 0.000 claims description 3
- 238000012935 Averaging Methods 0.000 claims description 2
- 230000001419 dependent effect Effects 0.000 claims 1
- 230000008569 process Effects 0.000 abstract description 4
- 206010002953 Aphonia Diseases 0.000 abstract 1
- 238000001514 detection method Methods 0.000 description 3
- 238000013179 statistical model Methods 0.000 description 3
- 238000004364 calculation method Methods 0.000 description 2
- 230000001934 delay Effects 0.000 description 2
- 238000010586 diagram Methods 0.000 description 2
- 230000004913 activation Effects 0.000 description 1
- 230000002238 attenuated effect Effects 0.000 description 1
- 230000000694 effects Effects 0.000 description 1
- 238000011156 evaluation Methods 0.000 description 1
- 238000001914 filtration Methods 0.000 description 1
- 238000009499 grossing Methods 0.000 description 1
- 238000009434 installation Methods 0.000 description 1
- 238000002955 isolation Methods 0.000 description 1
- 238000005259 measurement Methods 0.000 description 1
- 230000007246 mechanism Effects 0.000 description 1
- 230000002787 reinforcement Effects 0.000 description 1
- 230000009466 transformation 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
Definitions
- the present invention relates to the denoising of audio signals picked up by a microphone in a noisy environment.
- the invention is advantageously applied, but in a nonlimiting manner, to the speech signals picked up by the hands-free telephones or the like.
- These devices include a sensitive microphone not only capturing the voice of the user, but also the surrounding noise, noise that is a disruptive element that can go, in some cases, to make incomprehensible the speaker's words.
- the WO-A-98/45997 uses the push-button activation of a phone (for example when the driver wants to answer an incoming call) to detect the beginning of a speech signal and consider that the signal previously received to this support was essentially a noise signal. This last signal, stored, is analyzed to give a weighted average energy spectrum of the noise, then subtract from the noisy speech signal.
- the US-A-5,742,694 describes another technique, implementing a predictive adaptive filter type mechanism.
- This filter delivers a "reference signal” corresponding to the predictable part of the noisy signal and an "error signal” corresponding to the prediction error, then attenuates these two signals in variable proportions, and recombines them to provide a signal noised.
- Still other techniques called beamforming or double-phoning , implement two separate microphones.
- the first is designed and placed to primarily capture the speaker's voice, while the other is designed and placed to capture a larger noise component than the main microphone.
- the comparison of the signals captured makes it possible to extract the voice of the ambient noise efficiently, and by relatively simple software means.
- This technique based on a spatial coherence analysis of two signals, however, has the disadvantage of requiring two remote microphones, which generally confines it to fixed or semi-fixed installations and does not allow to integrate it into a pre-existing device by simply adding a software module. It also assumes that the speaker's position relative to the two microphones is approximately constant, which is generally the case in a car phone used by its driver. In addition, to achieve a near satisfactory denoising, the signals are subjected to a significant pre-filtering, which again has the disadvantage of introducing distortions that degrade the quality of the signal rendered denoised.
- the invention relates to a technique for denoising audio signals picked up by a single microphone recording a voice signal in a noisy environment.
- the application of a gain called gain LSA makes it possible to minimize the mean square distance between the logarithm of the amplitude of the estimated signal and the logarithm of the amplitude of the original speech signal.
- This second criterion is superior to the first because the distance chosen is much better suited to the behavior of the human ear and therefore qualitatively gives better results.
- the essential idea is to reduce the energy of the very noisy frequency components by applying a low gain while leaving intact (by the application of a gain equal to 1) those that are little or no at all.
- the knowledge of the indices of the frames where the speech is absent makes it possible to evaluate the power of the noise as well as its evolution over time on this segment of the spectrum. It suffices to measure the energy of the raw signal when the speech is absent and to make an average continuously updated these measurements. The main question is therefore when exactly the speech of the speaker is absent from the signal picked up by the microphone.
- the method described in this article is not intended to identify precisely on which frequency components of which frames the speech is absent, but rather to give a confidence index between 0 and 1, a value of 1 indicating that the speech is absent for sure (according to the algorithm) while a value 0 declares the opposite.
- this index is likened to the probability of absence of speech a priori , ie the probability that speech is absent on a given frequency component of the frame considered. This is of course a non-rigorous assimilation in the sense that even if the presence of speech is probabilistic ex ante, the signal picked up by the microphone can at any moment only go through two distinct states. It can either (at the moment considered) include speech or not contain it.
- One of the aims of the invention is to overcome the drawbacks of the methods proposed up to now, by means of an improved denoising method applicable to a speech signal considered in isolation, in particular a signal picked up by a single microphone, a method which is based on the analysis of the temporal coherence of the captured signals.
- the starting point of the invention lies in the observation that speech generally has a temporal coherence greater than noise and that, as a result, it is clearly more predictable.
- the invention proposes to use this property to calculate a reference signal where the speech has been more attenuated than the noise, by applying in particular a predictive algorithm which may for example be of the LMS ( Least Mean Squares, Least Mean Squares ) type. ).
- This reference signal derived from the speech signal to be denoised may be used in a manner comparable to that of the signal of the second microphone of beam-forming techniques. two-way, for example techniques similar to those of Cohen and Berdugo [4, supra].
- the calculation of a ratio between the respective energy levels of the original signal and the reference signal thus obtained will make it possible to discriminate between the speech components and the nonstationary noise noises, and will provide an estimate of the probability of presence of speech of independently of any statistical model.
- the technique proposed by the invention implements an "intelligent subtraction” implying, after a linear prediction made on the passed samples of the original signal (and not of a prefiltered signal, thus degraded), a registration phase between the original signal and the predicted signal.
- the technique of the invention turns out, in practice, sufficiently powerful to provide extremely effective denoising directly on the original signal, freeing distortions introduced by a prefiltering chain, become unnecessary.
- the predictive algorithm is advantageously a recursive adaptive algorithm of LMS mean least squares type.
- Step c) advantageously comprises the application of a variable gain algorithm depending on the probability of presence / absence of speech, in particular an OM-LSA optimized modified log-spectral amplitude gain type algorithm.
- the signal that we want to denoise is a sampled digital signal x (n) , where n denotes the number of the sample ( n is the temporal variable).
- the noisy signal x (n) is applied as input to a predictive LMS algorithm schematized by block 10, including the application of appropriate delays 12.
- a predictive LMS algorithm schematized by block 10 including the application of appropriate delays 12.
- the short-term Fourier transform of the captured signal x (n) (block 16) and the signal y (n) delivered by the predictive LMS algorithm (block 14) are then calculated. From these two transforms is calculated a reference signal (block 18), which is one of the input variables of an algorithm for calculating the probability of absence of speech (block 24). Meanwhile, the noisy signal transform x (n), from block 16, is also applied to the probability calculation algorithm.
- Blocks 20 and 22 estimate the pseudo-stationary noise of the reference signal and the noisy signal transform is estimated, and the result is also applied to the probability calculation algorithm.
- the result of the speech absence probability calculation, as well as the noisy signal transform, are inputted to an OM-LSA gain processing algorithm (block 26), the result of which is subjected to an inverse transformation of Fourier (block 28) to give an estimate of speech de-noiseed.
- the predictive algorithm LMS (block 10) is shown diagrammatically in FIG.
- ⁇ i ⁇ not + 1 ⁇ i not + 2 ⁇ ⁇ not ⁇ x ⁇ not - ⁇ - i + 1 ⁇ being a gain constant which makes it possible to adjust the speed and the stability of the adaptation.
- the respective signals x (n) and y (n) (noisy speech signal and linear prediction) are split into frames of identical lengths, and their short-term Fourier transform (denoted respectively X and Y ) is calculated for each frame.
- the algorithm predicts a 50% overlap between consecutive frames, and the samples are multiplied by the coefficients of the Hanning window so that the addition of even and odd fields corresponds to the signal of origin itself.
- E ⁇ Ref k ⁇ l 2 E ⁇ S k ⁇ l 2 ⁇ ⁇ S k + E ⁇ D t k ⁇ l 2 ⁇ ⁇ D t k + E ⁇ D ps k ⁇ l 2 ⁇ ⁇ D ps k or ⁇ S k ⁇ ⁇ D t k ⁇ ⁇ D ps k represent the attenuation on the reference signal of the three signals in each spectrum segment.
- S being a smoothed estimate of the instantaneous energy:
- M being an estimator of the pseudo-stationary energy, which can be obtained for example by a method MCRA ( Minima Controlled Recursive Averaging ) of the same type as that described by Cohen and Berdugo [5, supra] (however, several alternatives exist in the literature).
- L x and L Ref are transient detection thresholds.
- ⁇ min (k) and ⁇ m ax (k) are the upper and lower limits for each spectrum segment. These various parameters are chosen so as to correspond to typical situations, close to reality.
- the next step (corresponding to block 26 of FIG. 1) consists in operating the denoising itself (reinforcement of the speech component).
- the estimator just described will be applied to the statistical model described by Ephraim and Malah [2, supra], which assumes that the noise and speech in each spectrum segment are independent Gaussian processes of respective variances ⁇ x ( k, l) and ⁇ d (k, l) .
- This step may advantageously implement the OM-LSA gain algorithm ( Optimally Modified Log-Spectral Amplitude Gain ) described by Cohen and Berdugo [3, cited above].
- the G min gain in the absence of speech hypothesis is a lower limit for noise reduction, in order to limit the distortion of speech.
- the signal obtained at the end of this treatment is subjected to an inverse Fourier transform (block 28) to give the final estimate of the denoised speech.
- the algorithm of the present invention is particularly effective in noisy environments, parasitized by both mechanical noises, vibrations, etc. as well as by musical noises, characteristic situations encountered in the interior of a car. Spectrograms show that the attenuation of the noise is not only effective, but is done without significant distortion of speech after denoising.
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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)
- Soundproofing, Sound Blocking, And Sound Damping (AREA)
- Signal Processing Not Specific To The Method Of Recording And Reproducing (AREA)
- Noise Elimination (AREA)
- Compression, Expansion, Code Conversion, And Decoders (AREA)
Applications Claiming Priority (1)
Application Number | Priority Date | Filing Date | Title |
---|---|---|---|
FR0601822A FR2898209B1 (fr) | 2006-03-01 | 2006-03-01 | Procede de debruitage d'un signal audio |
Publications (2)
Publication Number | Publication Date |
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EP1830349A1 true EP1830349A1 (de) | 2007-09-05 |
EP1830349B1 EP1830349B1 (de) | 2011-11-30 |
Family
ID=36992693
Family Applications (1)
Application Number | Title | Priority Date | Filing Date |
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EP07290219A Active EP1830349B1 (de) | 2006-03-01 | 2007-02-21 | Verfahren zur Geräuschdämpfung eines Audiosignals |
Country Status (6)
Country | Link |
---|---|
US (1) | US7953596B2 (de) |
EP (1) | EP1830349B1 (de) |
AT (1) | ATE535905T1 (de) |
ES (1) | ES2378482T3 (de) |
FR (1) | FR2898209B1 (de) |
WO (1) | WO2007099222A1 (de) |
Cited By (3)
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EP2131357A1 (de) * | 2008-06-04 | 2009-12-09 | Parrot | Automatisches Kontrollsystem der Verstärkung eines Audiosignals in Abhängigkeit vom Umgebungslärm |
FR2950461A1 (fr) * | 2009-09-22 | 2011-03-25 | Parrot | Procede de filtrage optimise des bruits non stationnaires captes par un dispositif audio multi-microphone, notamment un dispositif telephonique "mains libres" pour vehicule automobile |
CN116644281A (zh) * | 2023-07-27 | 2023-08-25 | 东营市艾硕机械设备有限公司 | 一种游艇船体偏移检测方法 |
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US8949120B1 (en) | 2006-05-25 | 2015-02-03 | Audience, Inc. | Adaptive noise cancelation |
FR2908004B1 (fr) * | 2006-10-26 | 2008-12-12 | Parrot Sa | Circuit de reduction de l'echo acoustique pour un dispositif "mains libres"utilisable avec un telephone portable |
FR2908003B1 (fr) * | 2006-10-26 | 2009-04-03 | Parrot Sa | Procede de reduction de l'echo acoustique residuel apres supression d'echo dans un dispositif"mains libres" |
FR2908005B1 (fr) * | 2006-10-26 | 2009-04-03 | Parrot Sa | Circuit de reduction de l'echo acoustique pour un dispositif "mains libres"utilisable avec un telephone portable |
US8521530B1 (en) * | 2008-06-30 | 2013-08-27 | Audience, Inc. | System and method for enhancing a monaural audio signal |
EP2151820B1 (de) * | 2008-07-21 | 2011-10-19 | Siemens Medical Instruments Pte. Ltd. | Verfahren zur Vorspannungskompensation zwecks temporärer cepstraler Glättung von Spektralfilterverstärkungen |
EP2407965B1 (de) * | 2009-03-31 | 2012-12-12 | Huawei Technologies Co., Ltd. | Verfahren und einrichtung zur audiosignalentrauschung |
FR2945696B1 (fr) * | 2009-05-14 | 2012-02-24 | Parrot | Procede de selection d'un microphone parmi deux microphones ou plus, pour un systeme de traitement de la parole tel qu'un dispositif telephonique "mains libres" operant dans un environnement bruite. |
WO2010151183A1 (en) * | 2009-06-23 | 2010-12-29 | Telefonaktiebolaget L M Ericsson (Publ) | Method and an arrangement for a mobile telecommunications network |
FR2948484B1 (fr) | 2009-07-23 | 2011-07-29 | Parrot | Procede de filtrage des bruits lateraux non-stationnaires pour un dispositif audio multi-microphone, notamment un dispositif telephonique "mains libres" pour vehicule automobile |
KR101587844B1 (ko) * | 2009-08-26 | 2016-01-22 | 삼성전자주식회사 | 마이크로폰의 신호 보상 장치 및 그 방법 |
US8219394B2 (en) * | 2010-01-20 | 2012-07-10 | Microsoft Corporation | Adaptive ambient sound suppression and speech tracking |
US8798290B1 (en) | 2010-04-21 | 2014-08-05 | Audience, Inc. | Systems and methods for adaptive signal equalization |
DK2395506T3 (da) * | 2010-06-09 | 2012-09-10 | Siemens Medical Instr Pte Ltd | Fremgangsmåde og system til behandling af akustisk signal til undertrykkelse af interferens og støj i binaurale mikrofonkonfigurationer |
US20120245927A1 (en) * | 2011-03-21 | 2012-09-27 | On Semiconductor Trading Ltd. | System and method for monaural audio processing based preserving speech information |
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FR2974655B1 (fr) | 2011-04-26 | 2013-12-20 | Parrot | Combine audio micro/casque comprenant des moyens de debruitage d'un signal de parole proche, notamment pour un systeme de telephonie "mains libres". |
FR2976111B1 (fr) * | 2011-06-01 | 2013-07-05 | Parrot | Equipement audio comprenant des moyens de debruitage d'un signal de parole par filtrage a delai fractionnaire, notamment pour un systeme de telephonie "mains libres" |
FR2976710B1 (fr) * | 2011-06-20 | 2013-07-05 | Parrot | Procede de debruitage pour equipement audio multi-microphones, notamment pour un systeme de telephonie "mains libres" |
US8880393B2 (en) * | 2012-01-27 | 2014-11-04 | Mitsubishi Electric Research Laboratories, Inc. | Indirect model-based speech enhancement |
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FR3044197A1 (fr) | 2015-11-19 | 2017-05-26 | Parrot | Casque audio a controle actif de bruit, controle anti-occlusion et annulation de l'attenuation passive, en fonction de la presence ou de l'absence d'une activite vocale de l'utilisateur de casque. |
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JP2020144204A (ja) * | 2019-03-06 | 2020-09-10 | パナソニック インテレクチュアル プロパティ コーポレーション オブ アメリカPanasonic Intellectual Property Corporation of America | 信号処理装置及び信号処理方法 |
FR3113537B1 (fr) | 2020-08-19 | 2022-09-02 | Faurecia Clarion Electronics Europe | Procédé et dispositif électronique de réduction du bruit multicanale dans un signal audio comprenant une partie vocale, produit programme d’ordinateur associé |
CN112233688B (zh) * | 2020-09-24 | 2022-03-11 | 北京声智科技有限公司 | 音频降噪方法、装置、设备及介质 |
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- 2006-03-01 FR FR0601822A patent/FR2898209B1/fr not_active Expired - Fee Related
-
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- 2007-02-21 EP EP07290219A patent/EP1830349B1/de active Active
- 2007-02-21 ES ES07290219T patent/ES2378482T3/es active Active
- 2007-02-21 AT AT07290219T patent/ATE535905T1/de active
- 2007-02-26 US US11/710,613 patent/US7953596B2/en active Active
- 2007-02-27 WO PCT/FR2007/000347 patent/WO2007099222A1/fr active Application Filing
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Cited By (5)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
EP2131357A1 (de) * | 2008-06-04 | 2009-12-09 | Parrot | Automatisches Kontrollsystem der Verstärkung eines Audiosignals in Abhängigkeit vom Umgebungslärm |
FR2950461A1 (fr) * | 2009-09-22 | 2011-03-25 | Parrot | Procede de filtrage optimise des bruits non stationnaires captes par un dispositif audio multi-microphone, notamment un dispositif telephonique "mains libres" pour vehicule automobile |
US8195246B2 (en) | 2009-09-22 | 2012-06-05 | Parrot | Optimized method of filtering non-steady noise picked up by a multi-microphone audio device, in particular a “hands-free” telephone device for a motor vehicle |
CN116644281A (zh) * | 2023-07-27 | 2023-08-25 | 东营市艾硕机械设备有限公司 | 一种游艇船体偏移检测方法 |
CN116644281B (zh) * | 2023-07-27 | 2023-10-24 | 东营市艾硕机械设备有限公司 | 一种游艇船体偏移检测方法 |
Also Published As
Publication number | Publication date |
---|---|
ES2378482T3 (es) | 2012-04-13 |
US20070276660A1 (en) | 2007-11-29 |
ATE535905T1 (de) | 2011-12-15 |
FR2898209A1 (fr) | 2007-09-07 |
US7953596B2 (en) | 2011-05-31 |
EP1830349B1 (de) | 2011-11-30 |
FR2898209B1 (fr) | 2008-12-12 |
WO2007099222A1 (fr) | 2007-09-07 |
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