EP0642290A2 - Appareil de communication mobile avec dispositif pour le traitement de la parole - Google Patents

Appareil de communication mobile avec dispositif pour le traitement de la parole Download PDF

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Publication number
EP0642290A2
EP0642290A2 EP94202513A EP94202513A EP0642290A2 EP 0642290 A2 EP0642290 A2 EP 0642290A2 EP 94202513 A EP94202513 A EP 94202513A EP 94202513 A EP94202513 A EP 94202513A EP 0642290 A2 EP0642290 A2 EP 0642290A2
Authority
EP
European Patent Office
Prior art keywords
signal
microphone
speech
interference signal
microphone signals
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.)
Withdrawn
Application number
EP94202513A
Other languages
German (de)
English (en)
Other versions
EP0642290A3 (fr
Inventor
Walter Prof. Dr. Ing. C/O Philips Kellermann
Current Assignee (The listed assignees may be inaccurate. Google has not performed a legal analysis and makes no representation or warranty as to the accuracy of the list.)
Philips Intellectual Property and Standards GmbH
Koninklijke Philips NV
Original Assignee
Philips Corporate Intellectual Property GmbH
Philips Patentverwaltung GmbH
Koninklijke Philips Electronics NV
Philips Electronics NV
Priority date (The priority date 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 date listed.)
Filing date
Publication date
Application filed by Philips Corporate Intellectual Property GmbH, Philips Patentverwaltung GmbH, Koninklijke Philips Electronics NV, Philips Electronics NV filed Critical Philips Corporate Intellectual Property GmbH
Publication of EP0642290A2 publication Critical patent/EP0642290A2/fr
Publication of EP0642290A3 publication Critical patent/EP0642290A3/fr
Withdrawn legal-status Critical Current

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Classifications

    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; ELECTRIC HEARING AIDS; PUBLIC ADDRESS SYSTEMS
    • H04R3/00—Circuits for transducers
    • H04R3/005—Circuits for transducers for combining the signals of two or more microphones
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; ELECTRIC HEARING AIDS; PUBLIC ADDRESS SYSTEMS
    • H04R1/00—Details of transducers, loudspeakers or microphones
    • H04R1/20—Arrangements for obtaining desired frequency or directional characteristics
    • H04R1/32—Arrangements for obtaining desired frequency or directional characteristics for obtaining desired directional characteristic only
    • H04R1/40—Arrangements for obtaining desired frequency or directional characteristics for obtaining desired directional characteristic only by combining a number of identical transducers
    • H04R1/406—Arrangements for obtaining desired frequency or directional characteristics for obtaining desired directional characteristic only by combining a number of identical transducers microphones
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; ELECTRIC HEARING AIDS; PUBLIC ADDRESS SYSTEMS
    • H04R2430/00—Signal processing covered by H04R, not provided for in its groups
    • H04R2430/20—Processing of the output signals of the acoustic transducers of an array for obtaining a desired directivity characteristic
    • H04R2430/23—Direction finding using a sum-delay beam-former

Definitions

  • the invention relates to a mobile radio device with a speech processing device with at least two microphones, which are used to deliver microphone signals consisting of speech and interference signal components to microphone signal branches, which are coupled to the inputs of an adding device used to form a sum signal.
  • a microphone arrangement is known from four microphones located in the corners of a room with a square floor plan, the Microphone signals are further processed so that the influence of interference signals that are superimposed on speech signals is reduced.
  • the microphone signals are first shifted against each other in time in order to compensate for time differences between a speaker and the individual microphones.
  • the microphone signals with thus in-phase speech signal components are superimposed by an adding device to form a sum signal, so that the uncorrelated interference signal components of the microphone signals are weakened during the superimposition.
  • the attenuation is not optimal if there is an inhomogeneous interference signal field.
  • the superimposed microphone signals are fed to an adaptive filter (Wiener filter) by means of a correction factor serving to form the mean value. This is set by evaluating the in-phase microphone signals and further suppresses the interference signals.
  • the object of the invention is to improve the suppression of the interference signal component of the sum signal present at the output of the adding device.
  • the signal-to-noise ratio corresponds to the ratio of the powers of the speech and interference signal components of the sum signal.
  • the influence of an inhomogeneity of the interference signal field is minimized.
  • Microphone signals with small interfering signal components are compared to the microphone signals large interfering signal components amplified. Due to the correlated nature of the speech signals and the uncorrelated nature of the interference signals, this leads to the sum signal present at the output of the adding device having a reduced interference signal component or an increased signal / noise ratio, as a result of which the speech signal of the sum signal is better understood.
  • the computation of the weight factors which does not require much computation, leads to an increased signal / noise ratio and improved speech intelligibility. Because of the efficient calculation of the weighting factors, a calculation that is often required in speech processing is possible in real time, so that there is no annoying delay during a conversation conducted via the speech processing device.
  • an adaptation of the weighting factors to changes in the interference signal components over time is provided.
  • the weighting factors are kept constant in periods in which a satisfactory stationarity of the signal statistics of the interference signals is assumed. The length of these time segments depends on the nature of the respective interference signal field.
  • Another embodiment of the invention is characterized in that in each microphone signal branch a transformation device for spectral transformation of the assigned microphone signal, it is provided that the evaluation circuit is provided to form weight factors for each section of the spectral range of the microphone signals and that in each microphone signal branch a reverse transformation device is arranged after a means for weighting the spectral section sections.
  • the interference signal components of the microphone signals generally have no spectra with spectral values of the same size. For this reason, it makes sense to determine the weighting factors of the microphone signals and the weighting not in the time domain but in the spectral range, for which purpose a transformation of the microphone signals - for example with a Fourier transformation - is required.
  • the spectral range is divided into sections with at least one spectral value. For each spectral range section, the optimal weight factors are determined with which the corresponding spectral values of the microphone signals are weighted. An improved reduction of the interference signal components of the microphone signals is achieved and speech intelligibility is further increased.
  • x i stands for the microphone signal generated by the microphone M i , s i for the speech signal component contained therein and n i for the corresponding interference signal component in each case in the i-th microphone signal branch. The following designations apply to the digitized signals as for the corresponding analog signals.
  • the interference signals are normally noise signals that are caused, for example, by engine or wind noise when used in vehicles.
  • the outputs of the analog-digital converter 1 are connected to N inputs of a preprocessing unit 2. This contains for each microphone signal branch a delay element T1, ..., T N , whereby differences in transit time of speech signals from a speech signal source to the microphones M1, ..., M N are compensated.
  • the delay elements T1, ..., T N are adaptively adapted to these time differences.
  • the weight factors c1, ..., c N are set by an evaluation unit 4, which determines this by evaluating the microphone signals x1, ..., x N according to a scheme to be explained. If an approximate temporal steadiness of the statistical properties of the interference signal components n i can be assumed, a one-off calculation of the weighting factors is sufficient.
  • the filter 6 is set with the aid of the evaluation unit 4 by evaluating the microphone signals, for example as in the prior art cited at the beginning.
  • the estimates of the amplitudes of the speech signal components s i are obtained by Determination of difference determined.
  • the weighting factors c 1, ..., c N are to be dimensioned such that the so-called signal-to-noise ratio (SNR) of the sum signal x at the output of the adding device 5 is maximized.
  • SNR results from the ratio of the power (variance) of the speech signal component to the power (variance) of the interference signal component of the sum signal x.
  • ⁇ s and ⁇ n are the standard deviations of the speech signal component s and the interference signal component n of the sum signal x.
  • n1 thus serves as a reference interference signal. All other microphone signals or speech and interference signal components with an index i ⁇ 1 can also be set as reference variables without restriction.
  • Interference signal ratios b i 2 are thus defined by the ratio of the estimated powers ⁇ ni 2 of the interference signal components to the estimated power ⁇ n1 2 of the reference interference signal component.
  • the speech processing device described by FIGS. 2 and 3 represents an embodiment of the speech processing device shown in FIG. 1.
  • the N output signals of the preprocessing unit 2, which represent the samples of the microphone signals x 1,..., N are converted into spectral transformation devices 7 in the Spectral range transformed, for example by fast Fourier transform (FFT).
  • FFT fast Fourier transform
  • the spectral range is divided into M sections that contain at least one spectral value.
  • the spectral values are given to N multiplication devices 8, each section of the spectral range weighted or multiplied by a weight factor c i, j calculated separately for each spectral range section.
  • i is the index of the microphone signal branch.
  • j represents the spectral or frequency index of the respective spectral range section.
  • one of the multiplication devices 8 is shown in its basic structure, which multiplies the spectral range sections of the respective microphone signal branch by the weighting factors c i, j .
  • the spectral range contains M spectral range sections, so that M multipliers are required for each microphone signal branch.
  • the weighting factors c i, j are set by an evaluation unit 9. They are determined analogously to the calculation of the weighting factors c i in the description of FIG. 1 by maximizing the signal / noise ratio (SNR) in the respective spectral range sections.
  • SNR signal / noise ratio
  • the estimated values of the amplitudes of the speech and interference signal components s i , n i in the time domain are to be replaced by corresponding estimated values in the frequency domain.
  • the spectral values weighted in this way are fed back transformation devices 10, which transform the weighted spectra of the respective microphone signal branches back into the time domain.
  • the signals obtained in this way are added up as in FIG. 1 by the adding device 5 and fed to the adaptive filter 6.
  • This is set by an evaluation unit 11 which, analogous to the evaluation unit 4 in FIG. 1, evaluates the microphone signals x i present at the outputs of the analog-digital converter 1.
  • the signal-to-noise ratio (SNR) of the sum signal x can be further increased and speech intelligibility can be improved, since it is taken into account that the power of the interference signal components in the spectral range is not uniformly distributed over all spectral values.
  • the weighting factors c i and c i, j are constantly recalculated and set. This depends on the nature of the respective interference signal field. For example, the interference signal statistics of a vehicle change considerably when accelerating from a standing position, since noise is now generated, for example, by the headwind.
  • a mobile device 12 in which a voice processing device 13 is integrated, which are supplied via an arrangement of three microphones M1, M2 and M3 microphone signals.
  • the structure of the speech processing device 13 can be found either in FIG. 1 or in FIGS. 2 and 3 with the associated descriptions.
  • Output signals of the speech processing device 13 are fed to a function block 14, which combines the other functional units of the mobile radio device 12 and to which a loudspeaker 15 and an antenna 16 are coupled.
  • the microphones M 1, M 2 and M 3, the speech processing device 13 and the loudspeaker 15 act with the help of the function block 14 as parts of a hands-free device of the mobile radio device 12.

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  • Health & Medical Sciences (AREA)
  • Otolaryngology (AREA)
  • Physics & Mathematics (AREA)
  • Engineering & Computer Science (AREA)
  • Acoustics & Sound (AREA)
  • Signal Processing (AREA)
  • General Health & Medical Sciences (AREA)
  • Circuit For Audible Band Transducer (AREA)
  • Mobile Radio Communication Systems (AREA)
EP94202513A 1993-09-07 1994-09-02 Appareil de communication mobile avec dispositif pour le traitement de la parole Withdrawn EP0642290A3 (fr)

Applications Claiming Priority (2)

Application Number Priority Date Filing Date Title
DE4330243 1993-09-07
DE4330243A DE4330243A1 (de) 1993-09-07 1993-09-07 Sprachverarbeitungseinrichtung

Publications (2)

Publication Number Publication Date
EP0642290A2 true EP0642290A2 (fr) 1995-03-08
EP0642290A3 EP0642290A3 (fr) 2006-04-19

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Country Status (4)

Country Link
US (1) US5602962A (fr)
EP (1) EP0642290A3 (fr)
JP (1) JP3373306B2 (fr)
DE (1) DE4330243A1 (fr)

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JP3373306B2 (ja) 2003-02-04
EP0642290A3 (fr) 2006-04-19
US5602962A (en) 1997-02-11
JPH07240992A (ja) 1995-09-12
DE4330243A1 (de) 1995-03-09

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