EP1143416A2 - Suppression de bruit dans le domaine temporel - Google Patents

Suppression de bruit dans le domaine temporel Download PDF

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Publication number
EP1143416A2
EP1143416A2 EP01440083A EP01440083A EP1143416A2 EP 1143416 A2 EP1143416 A2 EP 1143416A2 EP 01440083 A EP01440083 A EP 01440083A EP 01440083 A EP01440083 A EP 01440083A EP 1143416 A2 EP1143416 A2 EP 1143416A2
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EP
European Patent Office
Prior art keywords
signal
frequency
noise
frequency spectrum
noise signal
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.)
Granted
Application number
EP01440083A
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German (de)
English (en)
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EP1143416B1 (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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Application filed by Alcatel CIT SA, Alcatel SA filed Critical Alcatel CIT SA
Publication of EP1143416A2 publication Critical patent/EP1143416A2/fr
Publication of EP1143416A3 publication Critical patent/EP1143416A3/fr
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Publication of EP1143416B1 publication Critical patent/EP1143416B1/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

  • a known method for noise reduction is the so-called “spectral Subtraction”, for example in the publication”
  • a new approach to noise reduction based on auditory masking effects "by S. Gustafsson and P. Jax, ITG conference, Dresden, 1998. It is about a spectral noise reduction method in which an acoustic Masking threshold (e.g. according to the MPEG standard) is taken into account.
  • Noise problems in newer applications are particularly severe of communication systems such as cell phones which the end devices are designed so small that an immediate spatial proximity between loudspeaker and microphone cannot be avoided is. Because of the direct sound transmission, especially through structure-borne noise The acoustic interference signal can be between the loudspeaker and the microphone come in the same order of magnitude as the speaker's useful signal on respective terminal or even exceed in amplitude. Such a thing Noise problem also occurs with several spatially adjacent Devices, for example in an office or conference room with many Telephone connections to a not inconsiderable extent, since a coupling of every loudspeaker signal on every microphone.
  • interference signals like unwanted background noise (Street noise, factory noise, office noise, canteen noise, aircraft noise etc.) or completely suppress.
  • the Compander thus consists of two sub-functions, a compressor for Speech signal levels that are greater than or equal to a normal level and one Expander for signal levels that are lower than the normal level.
  • the spectral subtraction mentioned above is used for this purpose first measured the noise during the pauses in the speech and in the form of a Power density spectrum continuously stored in a memory.
  • the power density spectrum is obtained via a Fourier transformation.
  • the stored noise spectrum is "the best current estimate "subtracted from the current disturbed speech spectrum, then transformed back into the time domain to create a Get noise reduction for the disturbed signal.
  • a disadvantage of such methods is the complex determination of these acoustic Concealment threshold and the execution of all with this procedure associated arithmetic operations.
  • Another disadvantage of spectral subtraction consists in the fact that the process is fundamentally not exact spectral noise estimation and subsequent subtraction also errors in the Output signal occur that make themselves felt as "musical tones”.
  • a spectral acoustic masking threshold R T (f) for the human ear is then calculated using, for example, the rules from the MPEG standard.
  • a filter pass curve H (f) is calculated according to a simple rule, which is designed in such a way that essential spectral parts of the speech are passed through as unchanged as possible and spectral parts of the noise are reduced as much as possible.
  • the object of the present invention is to develop a method as far as possible to introduce less complexity with the features described at the beginning, in which a noise reduction in a technically uncomplicated manner or noise suppression is achieved, and at which the original signal remains intact until the actual noise is extracted.
  • the procedure should be simple, especially with less computation than previously possible, one for the human ear if possible to create a pleasant overall acoustic impression, depending on your taste can be adapted to individual needs.
  • the new method is completely independent of the requirements for speech signal processing can be carried out and thus a simple optimization to the requirements of spectral processing of noise signals enable.
  • the method according to the invention enables the separate replication of the noise signal in the frequency domain regardless of processing a direct deduction of the simulated one from the original speech signal 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 method according to the invention is less complex than the known methods from the prior art described above, requires less computing power and leads to better frequency resolution.
  • step (d) By separating the noise simulation from the transmission of the original signal enables the inventive method 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. This can be used to carry out the method according to the invention further minimizes required computing power or the process itself be done even faster.
  • a further development of this method variant is characterized in that the selection of the to generate 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.
  • the value for the noise signal to be simulated is not only derived from the current power value of an original signal in speech pauses alone, but also from a weighted spectral curve of the corresponding signal determined and overall an aurally correct, i.e. achieves a psychoacoustically pleasant sound reduction.
  • Step (c) or before step (d) takes place By choosing a specific frequency Differences in signal energy from a frequency group are special easy to detect.
  • a variant of the method is also advantageous in which the frequency spectrum in step (b) of the branched TK signal only in a predetermined frequency range is produced. If the source of interference is only a limited frequency spectrum has significant computing power be saved. For example, in motor vehicles with sources of interference in a frequency range up to a maximum of 1 KHz, since the Interference signal mainly due to low-frequency sound generation (engine, Gear, rolling noise etc.) is formed.
  • a method variant is particularly simple, which is characterized in that a discrete Fourier transformation or an inverse discrete Fourier transformation is used in step (b) and / or in step (d), with discrete-time amplitude values of the incoming TK signal a sampling frequency f T can be sampled.
  • FFT fast Fourier transform
  • IDFT inverse discrete Fourier transform
  • the IDFT can therefore be advantageous for a defined frequency range be applied.
  • the distribution of frequencies can be individual respectively. From a frequency resolution of less than 128 frequency lines it is possible to save computing power compared to the FFT.
  • An alternative to the last-mentioned method variant is an embodiment in which only the part of the frequency spectrum generated that is below half the sampling frequency f T / 2 is selected. This in turn saves computing power, but also saves storage space.
  • a variant of the method according to the invention is also particularly advantageous, in which a frequency spectrum is temporarily stored in step (c) by averaging the frequency spectrum currently generated in step (b) previously generated frequency spectra is obtained. By averaging Spectral lines with high energy found and random values or sporadic Systematically suppressed errors.
  • the averaging has different relative values Weighting the currently generated frequency spectrum in different Frequency ranges.
  • Weighting the natural settling behavior of interferers are taken into account. For example, the speed of an engine in a motor vehicle usually do not change abruptly. Low-frequency interferers have one higher settling time than high-frequency.
  • the suggested weighting helps making the adaptivity of a system stable and fast.
  • step (e) one according to predetermined criteria with a weighting factor a ⁇ 1 weighted simulated noise signal from the currently arriving TK signal subtracted.
  • the weighting factor a is one of 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 will be automatic recorded, the weighting can also take place during operation.
  • the weighting factor a can be used as a by the user of the TK-Systems selectable quality measure, adjustable value can be selected.
  • a weighting factor defined by the user enables an individual, User-defined adaptation of the method according to the invention to the individual Needs.
  • the system according to the invention in an existing integrated overall concept can be provided by the user statistical value, such as the probability of error or detection rate can be used to control the weighting factor.
  • the weighting factor can also be derived from the speed or speed.
  • weighting factor a adaptive is adapted to the current incoming TC signal.
  • the adaptive weighting allows automatic optimization of noise reduction during of the company.
  • the weighting factor can depend on statistical values such as the probability of errors, Average, changes in state, etc. can be derived. Adaptive weighting makes it particularly easy and quick to make adjustments the inventive method to individual circumstances in the acoustic environment of the telecommunications terminal possible.
  • Another advantageous variant of the method according to the invention is distinguished is characterized in that the simulated noise signal generated in step (d) a synthetic noise signal is added before step (e).
  • the Adding an artificial noise signal with constant power density can be used to mask dynamic, non-stationary interferers in the output signal serve.
  • TK signal currently arriving before step (e) of a defined time delay is subjected, which is preferably designed so that the phase position of the incoming TK signal with the phase position of the simulated noise signal before deduction.
  • the currently arriving TK signal is immediately supplied to the deduction in step (e), and that the simulated noise signal is in phase before step (e) the phase position of the currently arriving TC signal is adjusted.
  • the Phase position of the reproduced noise signal in the frequency range before Corrected back transformation, the subtraction from the undelayed signal done in the time domain.
  • Annoying signal delays can thus be eliminated.
  • a variant of the method according to the invention is particularly preferred for which is present in addition to the detection and reduction of noise signals of echo signals is detected and / or predicted and the echo signals suppressed or reduced.
  • An additional echo cancellation is however only possible if the received original signal from a distant TC subscriber is involved in the echo calculation. This means that the sound reproduction also includes an echo production, which is associated with a incoming signal from the remote TC subscriber is connected.
  • This method variant can be improved in that the control the reduction of noise signals and the reduction of echo signals done separately.
  • the artificial noise signal can be a previously during the current TK connection include recorded sound signal that the current can reproduce the acoustic surrounding situation particularly "lifelike".
  • a server unit also falls within the scope of the present invention Processor assembly and a gate array assembly to support the The inventive method described above and a computer program to carry out the procedure.
  • the method can be used as a hardware circuit, as well as in the form of a computer program.
  • software programming for powerful DSP's preferred because new insights and additional functions are made easier by a Software changes can be implemented on existing hardware basis are.
  • methods can also be used as hardware components, for example in telecommunications end devices or telephone systems can be implemented.
  • FIG. 1 shows how an incoming original signal x, which contains a speech component s and a noise component n, is simulated on the one hand in a device 1 in the frequency domain in a device 1 and on the other hand the original signal X s + n is separated from the noise replication one Noise subtraction is supplied, with a time delay ⁇ optionally being possible.
  • the noise-reduced signal y s is then forwarded in the telecommunications system.
  • a speech pause detector which is practically always required for the simulation of noise 2 is provided, with which it is determined when the incoming Signal may contain voice signals or when there is a pause in speech.
  • the incoming TK signal of a Fourier transform FT subjected to the generation of a frequency spectrum and each of them resulting frequency spectrum is stored in a buffer 3.
  • the frequency spectra stored one after the other can be used with the help a device 4 are subjected to averaging.
  • the speech pause detector 2 determines that a speech pause has ended is and voice signals may also be present in the incoming original signal, becomes the last frequency spectrum stored in the buffer 3 (possibly averaged with previously recorded spectra) of an inverse Fourier transformation IFT subjected and in a subtractor 5 from Subtracted original signal, which was possibly subjected to a time delay ⁇ , to get a noise-free or at least noise-reduced signal.
  • IFT subjected and forwarded as a noise-reduced TC signal in the time domain. It takes place in the known methods in the prior art basically always a change in the original signal even before actual noise extraction instead.
  • 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 in blocks in the device 1b for noise simulation.
  • the time signal is subjected to a windowing (for example according to Hamming) in a corresponding upstream device 4 ′ or 4 ′′ before the transformation into the frequency range
  • Parallel processing is carried out in a further path with the same fenestration, only the signal being offset by half the window length and otherwise the noise signal to be simulated is calculated using the same means, as a result of which the errors generated by the fenestration can be compensated for.
  • the windowing is carried out in a device 4 'in the first path, then the time signal is subjected to a fast Fourier transformation FFT and the spectrum which arises is stored in a buffer 3'.
  • FFT fast Fourier transformation
  • An inverse fast Fourier transform IFFT follows each of the intermediate memories 3 ', 3 ", and the resulting spectra in the time domain are combined in an overlap device 6 to form a simulated noise signal Yn.
  • the simulated noise signal is then in turn subtracted by 5 optionally subtracted from the original signal X s + n by a time ⁇ 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 Another embodiment is shown in Fig. 5, where the branched incoming TC signal x s + n + e contains echo signals in addition to voice and noise signals.
  • an echo signal e is also input, which is further processed in a processing path parallel to the noise simulation 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 frequency spectrum obtained is buffered in a buffer 3a.
  • the echo signal e is also subjected to a windowing in a device 4b and then Fourier transformed.
  • the frequency spectra of both paths are temporarily stored in a buffer 3b and possibly subjected to averaging.
  • a fast inverse Fourier transformation IFFT then takes place separately on each of the two paths.
  • the simulated noise signal and the simulated echo signal overlap to form an overall signal y n + e to be subtracted, which is subtracted in the subtraction device 5 from the unchanged or delayed original signal x s + n + e by the noise - And to obtain echo-reduced TK signal y s .
  • the figures 6a to 6c finally show examples of noise signals in the frequency domain calculated by the method according to the invention.
  • the noise signal to be simulated was obtained from a fast Fourier transform FFT. You can see the typical mirror symmetry around half the frequency value f s / 2.
  • 6c finally shows the result of using a modified discrete Fourier transform with higher resolution, again only half of the frequency spectrum being processed up to the frequency f s / 2.

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  • Engineering & Computer Science (AREA)
  • Acoustics & Sound (AREA)
  • Audiology, Speech & Language Pathology (AREA)
  • Multimedia (AREA)
  • Health & Medical Sciences (AREA)
  • Computational Linguistics (AREA)
  • Human Computer Interaction (AREA)
  • Physics & Mathematics (AREA)
  • Quality & Reliability (AREA)
  • Signal Processing (AREA)
  • Soundproofing, Sound Blocking, And Sound Damping (AREA)
  • Noise Elimination (AREA)
  • Measurement Of Mechanical Vibrations Or Ultrasonic Waves (AREA)
  • Compression, Expansion, Code Conversion, And Decoders (AREA)
  • Mobile Radio Communication Systems (AREA)
  • Details Of Television Scanning (AREA)
  • Diaphragms For Electromechanical Transducers (AREA)
  • Plural Heterocyclic Compounds (AREA)
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
DE10017646 2000-04-08
DE10017646A DE10017646A1 (de) 2000-04-08 2000-04-08 Geräuschunterdrückung im Zeitbereich

Publications (3)

Publication Number Publication Date
EP1143416A2 true EP1143416A2 (fr) 2001-10-10
EP1143416A3 EP1143416A3 (fr) 2004-04-21
EP1143416B1 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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Cited By (3)

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Publication number Priority date Publication date Assignee Title
WO2013140070A1 (fr) * 2012-03-22 2013-09-26 Bodysens Procédé, terminal et casque de communication vocale full-duplex sans fil avec auto-synchronisation sans maître ni base
FR2988549A1 (fr) * 2012-03-22 2013-09-27 Bodysens Procede, terminal et casque de communication vocale sans fil avec auto-synchronisation
US9479278B2 (en) 2012-03-22 2016-10-25 Bodysens Method, terminal and headset for wireless full-duplex voice communication with auto-sync without base or sync-master

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Publication number Publication date
US20010028713A1 (en) 2001-10-11
CN1325222A (zh) 2001-12-05
US6801889B2 (en) 2004-10-05
AU3336101A (en) 2001-10-11
HU0101288D0 (en) 2001-06-28
DE50108051D1 (de) 2005-12-22
EP1143416B1 (fr) 2005-11-16
JP2001350498A (ja) 2001-12-21
HUP0101288A2 (hu) 2001-12-28
ATE310305T1 (de) 2005-12-15
DE10017646A1 (de) 2001-10-11
EP1143416A3 (fr) 2004-04-21
CN1225104C (zh) 2005-10-26

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