EP1143416B1 - Suppression de bruit dans le domaine temporel - Google Patents

Suppression de bruit dans le domaine temporel Download PDF

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Publication number
EP1143416B1
EP1143416B1 EP01440083A EP01440083A EP1143416B1 EP 1143416 B1 EP1143416 B1 EP 1143416B1 EP 01440083 A EP01440083 A EP 01440083A EP 01440083 A EP01440083 A EP 01440083A EP 1143416 B1 EP1143416 B1 EP 1143416B1
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EP
European Patent Office
Prior art keywords
signal
frequency
noise
frequency spectrum
process according
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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
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EP01440083A
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German (de)
English (en)
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EP1143416A2 (fr
EP1143416A3 (fr
Inventor
Michael Walker
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Alcatel CIT SA
Alcatel Lucent SAS
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Alcatel CIT SA
Alcatel SA
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Publication of EP1143416A2 publication Critical patent/EP1143416A2/fr
Publication of EP1143416A3 publication Critical patent/EP1143416A3/fr
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    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L21/00—Speech or voice signal processing techniques to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility
    • G10L21/02—Speech enhancement, e.g. noise reduction or echo cancellation
    • G10L21/0208—Noise filtering
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L21/00—Speech or voice signal processing techniques to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility
    • G10L21/02—Speech enhancement, e.g. noise reduction or echo cancellation
    • G10L21/0208—Noise filtering
    • G10L21/0216—Noise filtering characterised by the method used for estimating noise
    • G10L2021/02168—Noise filtering characterised by the method used for estimating noise the estimation exclusively taking place during speech pauses

Definitions

  • the call partners are not in the same acoustic environment and therefore are not aware of the acoustic situation at the other party's location. Therefore, a problem arises more intensively when one of the partners is forced to speak very loudly due to its acoustic environment, while the other partner generates low amplitude voice signals in a quiet acoustic environment.
  • the terminals are designed so small that an immediate spatial proximity between speaker and microphone unavoidable is. Due to the direct sound transmission, in particular by structure-borne noise between speaker and microphone, the acoustic interference signal come in the same order of magnitude as the useful signal of the speaker on respective terminal or this even exceed in amplitude. Such a thing Noise problem also occurs with several spatially adjacent Terminals, for example in an office or conference room with many Telephone connections in a significant extent, since a coupling of every loudspeaker signal is sent to each microphone.
  • noise such as unwanted background noise (Street noise, factory noise, office noise, canteen noise, aircraft noise, etc.) to reduce or completely suppress.
  • the degree of noise reduction according to a fixed predetermined transfer function has first the property of voice signals with a specific (pre-set) "normal speech signal level" (possibly called normal volume) practical unchanged from its input to the output. But now once the input signal is too loud, e.g. because a speaker is too close Microphone comes, so limits a dynamic compressor to the output level almost the same value as normally, adding the current gain in the Kompander is linearly lowered with increasing input volume. By this property remains the language at the output of the compander system about the same - no matter how strong the input volume fluctuates.
  • normal speech signal level possibly called normal volume
  • a signal with a level that is smaller than the Normal level is given to the input of the compander, so will that Signal is additionally attenuated by the gain being back-regulated to If possible, transmit background noise only attenuated.
  • the Kompander thus consists of two sub-functions, a compressor for Speech signal levels greater than or equal to a normal level and one Expander for signal levels lower than the normal level.
  • the noise measured in the speech pauses and in the form of a Power density spectrum continuously stored in a memory.
  • the power density spectrum is won over a Fourier transformation.
  • the stored sound spectrum "as the best current estimate "subtracted from the current disturbed speech spectrum, then transformed back into the time domain to create in this way a To obtain noise reduction for the disturbed signal.
  • a disadvantage of such methods is the complex determination of this acoustic Masking threshold and the execution of all with this procedure connected arithmetic operations.
  • Another disadvantage of the spectral subtraction is that by the process of a basically inaccurate spectral noise estimation and subsequent subtraction also errors in the Output signal occur, which are noticeable as "musical tones".
  • a spectral acoustic masking threshold R T (f) for the human ear is then calculated using the rules from the MPEG standard, for example.
  • a filter pass curve H (f) is calculated according to a simple rule, designed to let the essential spectral parts of the speech pass as unaltered as possible and reduce the spectral parts of the noise as much as possible.
  • Object of the present invention is in contrast, a method possible low complexity with the features described above, in a technically inexpensive way a noise reduction or noise suppression is achieved, and at the original signal remains untouched until the actual noise deduction.
  • the procedure should be simple, especially with less computational effort as far as possible, one for the human ear possible pleasant overall acoustic impression, depending on the taste can be adapted to individual needs.
  • the new method completely independent of the requirements for a voice signal processing can be performed and thus a simple optimization to the requirements of spectral processing of noise signals enable.
  • the inventive method made possible by the separate replica the noise signal in the frequency domain regardless of processing the original voice signal direct deduction of the replicated Noise signal from the original, unadulterated input signal, which neither a Fourier transform nor an inverse Fourier transform is subjected. With a corresponding phase correction in the frequency domain is even a noise subtraction from the original signal with no time delay possible.
  • the inventive method is less complex as the above-described known prior art methods, requires less computing power and leads to better frequency resolution.
  • step (d) By separating the noise simulation from the transmission of the original signal allows the process of the invention in a particularly preferred Variant that in step (d) only a selected part of the generated Frequency spectrum used to generate the simulated noise signal becomes.
  • the for carrying out the method according to the invention required computing power further minimized or the process itself be done even faster.
  • a development of this variant of the method is characterized in that the selection of the for generating the simulated noise signal used part of the frequency spectrum according to criteria of psychoacoustics according to the mean values of the perceptual spectrum of the human Hearing takes place.
  • the value for the sound signal to be reproduced is not only from the instantaneous power value of an original signal in speech pauses alone, but also from a weighted spectral course of the corresponding signal determined and in total over the function thus gained a hearing-correct, i.e. Achieved a psychoacoustically pleasing-sounding noise reduction.
  • the Selection of the signal used to generate the simulated noise signal Part of the frequency spectrum such that only discrete frequencies of the Spectrum are considered, and that the distance of the discrete frequencies steadily larger in the direction of higher frequencies, preferably after one logarithmic function is selected. This is the frequency resolution to the Perception of the human ear better adapted.
  • Step (c) or before step (d) takes place.
  • step (b) the frequency spectrum in step (b) the branched TK signal only in a predetermined frequency range is produced. If the source of interference only a limited frequency spectrum can, in turn, with this measure, considerable computing power be saved. For example, in motor vehicles with sources of interference in a frequency range only up to a maximum of 1 KHz to be expected, since the Interference signal mainly due to low-frequency sound generation (engine, Gear, rolling noise, etc.) is formed.
  • step (b) a discrete Fourier transform or an inverse discrete Fourier transform is applied, wherein the incoming TK signal with discrete time amplitude values a sampling frequency f T are sampled.
  • FFT fast Fourier transform
  • IDFT inverse discrete Fourier transform
  • step (b) broadband interferers can be processed particularly economically become.
  • an embodiment is selected in which only that part of the generated frequency spectrum which is below half the sampling frequency f T / 2. This in turn can be achieved in savings in computing power, but also in storage space.
  • step (c) a frequency spectrum is buffered, the by averaging the frequency spectrum currently generated in step (b) previously generated frequency spectra is obtained.
  • psychoacoustic Weighting the frequency dependent settling times to the adapted to human hearing. This achieves an optimization of the system in terms of naturalness, stability and adaptation time.
  • step (e) according to predetermined criteria with a weighting factor a ⁇ 1 weighted simulated noise signal from the currently arriving one TK signal deducted.
  • the weighting factor a is used as one of Faults of the TK system dependent constant value selected. This makes possible an inexpensive and simple optimization of the invention Procedure to the errors of the respective telecommunications system. The errors become automatic recorded, the weighting can also take place during operation.
  • the weighting factor a may be determined as one after one by the user of the TK-Systems selectable quality level adjustable value can be selected.
  • One such user-defined weighting factor allows an individual, custom adaptation of the method according to the invention to the individual Needs. Is the system of the invention in an existing integrated parent concept can be a user-provided statistical value, such as the error probability or recognition rate used to control the weighting factor.
  • the weighting factor for example also be derived from the speed or speed.
  • the weighting factor a is adaptive is adapted to the current incoming TK signal.
  • the adaptive weighting allows automatic optimization of noise reduction during of the operation.
  • the weighting factor may vary from statistical values such as probability of error, Mean value, state changes, etc. are derived. With the adaptive weighting are particularly easy and quick adjustments of the inventive method to individual circumstances in the acoustic environment of the telecommunications terminal possible.
  • a further advantageous variant of the method according to the invention is characterized characterized in that the simulated noise signal generated in step (d) before step (e) a synthetic noise signal is added.
  • the Admixture of an artificial noise signal with constant power density can mask dynamic, non-stationary interferers in the output signal serve.
  • a further variant of the method according to the invention provides that the currently arriving TK signal before step (e) of a defined time delay is subjected, which is preferably designed so that the phase angle of the incoming TK signal with the phase position of the simulated noise signal matches before withdrawal.
  • the currently arriving TK signal is fed without delay to the trigger in step (e), and that the simulated noise signal in its phase position before step (e) to the phase angle of the currently arriving TK signal is adjusted.
  • the Phase angle of the reproduced noise signal in the frequency range before Corrected inverse transformation the subtraction from the instantaneous signal take place in the time domain. Disturbing signal delays can thus be dispensed with. These inevitably occur in all procedures in which the useful signal (Language) makes the detour via two transformations, such as in the known spectral subtraction discussed above.
  • the noise reproduction also includes a echo reproduction that with a connected to the remote TK subscriber signal is connected.
  • This process variant can be improved by the fact that the control the reduction of noise signals and the reduction of echo signals done separately.
  • the scope of the present invention also includes a server line, a Processor assembly and a gate array assembly to support the method described above and a computer program to carry out the process.
  • the method can be used both as a hardware circuit, as well as in the form of a computer program.
  • software programming for powerful DSP's preferred because new knowledge and additional functions easier by a Modification of the software can be implemented on an existing hardware basis are.
  • methods can also be used as hardware components, for example in TK terminals or telephone systems are implemented.
  • Fig. 1 it is shown how from an incoming original signal x, which contains a voice portion s and a noise n, on the one hand in a device 1, a noise signal y n in the frequency domain is simulated and on the other hand, the original signal x s + n separated from the noise simulation of a Noise subtraction is supplied, wherein optionally a time delay time delay ⁇ can be made.
  • the noise-reduced signal y s is then forwarded in the TK system.
  • a simple embodiment is shown in which in the device 1 a for noise simulation a virtually always required speech pause detector 2 is provided, which determines when the incoming Signal may contain speech signals or when there is a speech break.
  • the incoming TK signal of a Fourier transform FT subjected to generating a Frequenzsprektrums and each of them resulting frequency spectrum stored in a buffer 3.
  • the time sequentially stored frequency spectra can help with a means 4 are averaged.
  • the speech pause detector 2 determines that a speech pause is over is and in the incoming original signal and speech signals can be present, becomes the last stored in the buffer memory 3 frequency spectrum (possibly averaged with previously recorded spectra) of an inverse Fourier transformation IFT subjected and in a Sub Volumeglied 5 of Original signal, which was possibly subjected to a time delay ⁇ deducted, to get a noise-free or at least noise-reduced signal.
  • FIG. 4 shows a further embodiment of the invention, in which the original signal x s + n, which is initially received in the time domain, is processed block-by-block in the device 1 b for noise simulation.
  • the time signal before the transformation into the frequency range is subjected to a windowing (eg according to Hamming) in a correspondingly upstream device 4 'or 4 "In order to compensate for the errors caused by the windowing during the inverse transformation, in addition to the processing in a first Path is made parallel processing in another path with the same fenestration, wherein only the signal is offset by half the window length and otherwise the simulated noise signal is calculated by the same means, whereby a compensation of the errors generated by the fenestration can be achieved.
  • a windowing eg according to Hamming
  • the windowing is performed in a device 4 ', then the time signal is subjected to a fast Fourier transformation FFT and the resulting spectrum is stored in an intermediate memory 3'.
  • FFT fast Fourier transformation
  • An inverse fast Fourier transformation IFFT is connected to the latches 3 ', 3 ", and the resulting spectra in the time domain are combined to form a simulated noise signal Yn in an overlap device 6.
  • the simulated noise signal in the subtraction element 5 is converted by a optionally subtracted by a time ⁇ time-offset original signal x s + n in order to obtain the noise-corrected output signal y S.
  • the subtraction of the noise signal from the original signal in the subtraction element 5 can be phase-adjusted.
  • FIG. 5 A further embodiment is shown in Fig. 5, where the branched incoming TK signal x s + n + e in addition to speech and noise signals also contains echo signals.
  • a device 1c for noise and echo replica also an echo signal e is input, which is further treated in a processing path parallel to the noise training path.
  • the incoming original signal X s + n + e is first subjected to a windowing in a device 4a, then a fast Fourier transform FFT and the obtained frequency spectrum are buffered in a buffer 3a.
  • the echo signal e in a device 4b is also subjected to a windowing and then Fourier-transformed.
  • the frequency spectra of both paths are buffered in a buffer 3b and possibly subjected to averaging. Thereafter, a fast inverse Fourier transformation IFFT is performed separately on both paths.
  • the simulated noise signal and the simulated echo signal are overlapped into a total signal y n + e to be subtracted, which is subtracted in the subtraction device 5 from the original signal x s + n + e delayed or delayed by a time ⁇ in order to record the noise and echo-reduced TK signal y s .
  • FIGS. 6a to 6c show examples of noise signals calculated in the frequency domain according to the method of the invention.
  • the noise signal to be reproduced has been obtained from a fast Fourier transformation FFT.
  • the typical mirror symmetry can be seen around half the frequency value f s / 2.
  • Fig. 6c shows the result of using a modified discrete Fourier transform with higher resolution, again processing only half of the frequency spectrum up to the frequency f s / 2.

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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)
  • Noise Elimination (AREA)
  • Mobile Radio Communication Systems (AREA)
  • Plural Heterocyclic Compounds (AREA)
  • Details Of Television Scanning (AREA)
  • Diaphragms For Electromechanical Transducers (AREA)
  • Compression, Expansion, Code Conversion, And Decoders (AREA)
  • Measurement Of Mechanical Vibrations Or Ultrasonic Waves (AREA)

Claims (14)

  1. Procédé permettant de réduire les signaux acoustiques dans les systèmes de télécommunication (=TK) pour la transmission de signaux utiles, en particulier, du langage humain, avec les étapes suivantes :
    (a) détermination au moyen d'une détection d'interruption de conversation, du moment où un signal vocal est contenu dans le mélange à transmettre constitué des signaux utiles et des signaux parasites ou du moment où une interruption de conversation est disponible ;
    (b) déviation du signal de télécommunication arrivant de la voie de signal principal et application d'une transformée de Fourier au signal de télécommunication dévié pour produire un spectre de fréquences du signal de télécommunication dévié ;
    (c) mémorisation du dernier spectre de fréquences reçu pendant la dernière interruption de conversation dans une mémoire tampon (3) ;
    (d) application d'une transformée de Fourier inverse au dernier spectre de fréquences reçu respectivement pour produire un signal acoustique simulé;
    (e) soustraction du signal acoustique simulé dans le domaine temporel du signal de télécommunication arrivant actuellement.
  2. Procédé selon la revendication 1, caractérisé en ce que à l'étape (d), seule une partie choisie du spectre de fréquences produit est utilisée pour produire le signal acoustique simulé.
  3. Procédé selon la revendication 2, caractérisé en ce que le choix du signal acoustique simulé utilisé pour produire la partie du spectre de fréquences est effectué d'après les critères de la psychoacoustique conformément aux moyennes du spectre de perception de l'audition humaine.
  4. Procédé selon la revendication 2, caractérisé en ce que le choix la partie du spectre utilisée pour produire le signal acoustique simulé est effectué de telle sorte que seules les fréquences discrètes du spectre sont considérées, et en ce que la distance des fréquences discrètes en direction des fréquences plus élevées est choisie constamment plus grande, de préférence, d'après une fonction logarithmique.
  5. Procédé selon la revendication 2, caractérisé en ce que la partie choisie du spectre de fréquences est répartie en groupes de fréquences fixés d'avance, et en ce que dans chaque groupe de fréquences seule la fréquence et/ou la bande de fréquences ayant la plus grande énergie de signal est choisie dans le groupe de fréquences et réutilisée pour produire le signal acoustique simulé.
  6. Procédé selon la revendication 5, caractérisé en ce que la sélection de la fréquence ou de la bande de fréquences est effectuée avec l'énergie de signal la plus grande à l'intérieur du groupe de fréquences avant l'étape (c) ou avant l'étape (d).
  7. Procédé selon la revendication 1, caractérisé en ce que à l'étape (b), le spectre de fréquences du signal de télécommunication dévié n'est produit que dans une gamme de fréquences prescrite.
  8. Procédé selon la revendication 1, caractérisé en ce que à l'étape (c) un spectre de fréquences est mémorisé dans la mémoire tampon, lequel spectre est obtenu par un établissement d'une moyenne du spectre de fréquences produit actuellement à l'étape (b) avec des spectres de fréquences produits auparavant.
  9. Procédé selon la revendication 8, caractérisé en ce que l'établissement d'une moyenne avec la pondération relative différente du spectre de fréquences produit actuellement est effectué dans des gammes de fréquences différentes.
  10. Procédé selon la revendication 9, caractérisé en ce que la pondération est effectuée d'après des critères de la psychoacoustique conformément aux moyennes du spectre de perception de l'audition humaine.
  11. Procédé selon la revendication 1, caractérisé en ce que à l'étape (e), un signal acoustique simulé pondéré d'après des critères prescrits avec un facteur de pondération a < 1 est soustrait du signal de télécommunication arrivant actuellement.
  12. Procédé selon la revendication 1, caractérisé en ce qu'un signal acoustique synthétique est ajouté au signal acoustique simulé produit à l'étape (d) avant l'étape (e).
  13. Procédé selon la revendication 1, caractérisé en ce que le signal de télécommunication arrivant actuellement est soumis avant l'étape (e) à un retard de temps défini qui est, de préférence, placé de telle sorte que la situation de phase du signal de télécommunication arrivant corresponde à la situation de phase du signal acoustique simulé avant la soustraction.
  14. Procédé selon la revendication 1, caractérisé en ce que le signal de télécommunication arrivant actuellement est conduit instantanément à la soustraction à l'étape (e), et en ce que le signal acoustique simulé est adapté dans sa situation de phase avant l'étape (e) à la situation de phase du signal de télécommunication arrivant actuellement.
EP01440083A 2000-04-08 2001-03-22 Suppression de bruit dans le domaine temporel Expired - Lifetime EP1143416B1 (fr)

Applications Claiming Priority (2)

Application Number Priority Date Filing Date Title
DE10017646A DE10017646A1 (de) 2000-04-08 2000-04-08 Geräuschunterdrückung im Zeitbereich
DE10017646 2000-04-08

Publications (3)

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EP1143416A2 EP1143416A2 (fr) 2001-10-10
EP1143416A3 EP1143416A3 (fr) 2004-04-21
EP1143416B1 true EP1143416B1 (fr) 2005-11-16

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US (1) US6801889B2 (fr)
EP (1) EP1143416B1 (fr)
JP (1) JP2001350498A (fr)
CN (1) CN1225104C (fr)
AT (1) ATE310305T1 (fr)
AU (1) AU3336101A (fr)
DE (2) DE10017646A1 (fr)
HU (1) HUP0101288A2 (fr)

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ATE310305T1 (de) 2005-12-15
HU0101288D0 (en) 2001-06-28
CN1225104C (zh) 2005-10-26
US20010028713A1 (en) 2001-10-11
EP1143416A2 (fr) 2001-10-10
DE10017646A1 (de) 2001-10-11
DE50108051D1 (de) 2005-12-22
EP1143416A3 (fr) 2004-04-21
US6801889B2 (en) 2004-10-05
JP2001350498A (ja) 2001-12-21
CN1325222A (zh) 2001-12-05
AU3336101A (en) 2001-10-11
HUP0101288A2 (hu) 2001-12-28

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