EP1892703B1 - Procédé et système fournissant un signal acoustique avec une largeur de bande étendue - Google Patents

Procédé et système fournissant un signal acoustique avec une largeur de bande étendue Download PDF

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EP1892703B1
EP1892703B1 EP06017456A EP06017456A EP1892703B1 EP 1892703 B1 EP1892703 B1 EP 1892703B1 EP 06017456 A EP06017456 A EP 06017456A EP 06017456 A EP06017456 A EP 06017456A EP 1892703 B1 EP1892703 B1 EP 1892703B1
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Prior art keywords
signal
broadband
bandwidth
bandwidth limit
acoustic signal
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German (de)
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EP1892703A1 (fr
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Tim Haulick
Bernd Iser
Gerhard Schmidt
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Harman Becker Automotive Systems GmbH
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Harman Becker Automotive Systems GmbH
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Priority to AT06017456T priority Critical patent/ATE446572T1/de
Priority to DE602006009927T priority patent/DE602006009927D1/de
Priority to EP06017456A priority patent/EP1892703B1/fr
Priority to CA002596411A priority patent/CA2596411A1/fr
Priority to JP2007214930A priority patent/JP5150165B2/ja
Priority to CN2007101466102A priority patent/CN101141533B/zh
Priority to KR1020070084306A priority patent/KR101433833B1/ko
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    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/02Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using spectral analysis, e.g. transform vocoders or subband vocoders
    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; SPEECH OR AUDIO CODING OR DECODING
    • G10L21/00Processing 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
    • G10L21/02Speech enhancement, e.g. noise reduction or echo cancellation
    • G10L21/038Speech enhancement, e.g. noise reduction or echo cancellation using band spreading techniques
    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/04Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using predictive techniques
    • G10L19/06Determination or coding of the spectral characteristics, e.g. of the short-term prediction coefficients
    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/04Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using predictive techniques
    • G10L19/08Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters
    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L2019/0001Codebooks

Definitions

  • the invention is directed to a method and a system for providing an acoustic signal, in particular a speech signal, with extended bandwidth.
  • Acoustic signals transmitted via an analog or digital signal path usually suffer from the drawback that the signal path only has a restricted bandwidth such that the transmitted acoustic signal differs considerably from the original signal. For example, in the case of conventional telephone connections, a sampling rate of 8 kHz is used resulting in a maximal signal bandwidth of 4 kHz. Compared to the case of audio CD's, the speech and audio quality is significantly reduced.
  • the bandwidth of telephone connections could be increased by using broadband or wideband digital coding and decoding methods (so-called broadband codecs).
  • broadband codecs wideband digital coding and decoding methods
  • both the transmitter and the receiver have to support corresponding coding and decoding methods which would require the implementation of a new standard.
  • systems for bandwidth extension can be used as described, for example, in P. Jax, Enhancement of Bandlimited Speech Signals: Algorithms and Theoretical Bounds, Dissertation, Aachen, Germany, 2002 or E. Larsen, R. M. Aarts, Audio Bandwidth Extension, Wiley, Hoboken, NJ, USA, 2004 .
  • These systems are to be implemented on the receiver's side only such that existing telephone connections do not have to be changed.
  • the missing frequency components of an input signal with small bandwidth are estimated and added to the input signal.
  • Fig. 6 An example of the structure and the corresponding signal flow in such a state of the art bandwidth extension system is illustrated in Fig. 6 .
  • both the lower and the upper frequency ranges are re-synthesized.
  • an incoming or received acoustic signal x ( n ) in digitized form is processed by sub-sampling and block extraction so as to obtain signal vectors x ( n ).
  • the variable n denotes the time.
  • the bandwidth extension is performed only within the missing frequency ranges.
  • the extension concerns low frequency (for example from 0 to 300 Hz) and/or high frequency (for example 3400 Hz to half of the desired sampling rate) ranges.
  • a narrowband spectral envelope is extracted from the narrowband signal, the narrowband signal being restricted by the bandwidth restrictions of the telephone channel.
  • a corresponding broadband envelope signal is estimated from the narrowband envelope.
  • the mappings are based, for example, on codebook pairs (see J. Epps, W. H. Holmes, A New Technique for Wideband Enhancement of Coded Narrowband Speech, IEEE Workshop on Speech Coding, Conference proceedings, pages 174 to 176 June 1999 ) or on Neural Networks (see J.-M. Valin R. Lefebvre, Bandwidth Extension of Narrowband Speech for Low Bit-Rate Wideband Coding, IEEE Workshop on Speech Coding, Conference Proceedings, pages 130 to 132, September 2000 ). In these methods, the entries of the codebooks or the weights of the neural networks are generated using training methods requiring large processor and memory resources.
  • a broadband or wideband excitation signal having a spectrally flat envelope is generated from the narrowband signal.
  • This excitation signal corresponds to the signal which would be recorded directly behind the vocal cords, i.e. the excitation signal contains information about voicing and pitch, but not about form and structures or the spectral shaping in general.
  • the excitation signal has to be weighted with the spectral envelope.
  • non-linear characteristics see U. Kornagel, Spectral Widening of the Excitation Signal for Telephone-Band Speech Enhancement, IWAENC 01, Conference Proceedings, pages 215 to 218, September 2001
  • two-ray rectifying or squaring for example.
  • the excitation signal x exc ( n ) is spectrally colored using the envelope in block 604.
  • the spectral ranges used for the extension are extracted using a band stop filter in block 606 resulting in signal vectors y ext ( n ).
  • the band stop filter can be effective, for example, in the range from 200 to 3700 Hz.
  • the signal vectors x(n) of the received signal are passed through a complementary band pass filter in block 605. Then, the signal components y ext ( n ) and y tel ( n ) are added to obtain a signal vector y ( n ) with extended bandwidth. In block 607, the different signal vectors are assembled again and an over-sampling is performed resulting in a signal y (n).
  • EP 0 944 036 discloses a method and a device for detecting voice sections.
  • a method for speech bandwidth extension is disclosed in US 2002/0138268 .
  • a frequency interpolating device and method is known from EP 1 298 643 .
  • a method for providing an acoustic signal with extended bandwidth comprising:
  • the method according to the invention allows an adaptation of the bandwidth extension to the acoustic signal actually received. For example, when the transmitter uses an ISDN telephone, a broader frequency range is used compared to the case of a mobile phone with a hands-free system. Therefore, the bandwidth of a received acoustic signal will be extended only in those ranges where it is necessary so that the quality of the resulting signal is very high.
  • the received acoustic signal may be a digital signal or may be digitized.
  • steps (a) to (c) may be preceded by the step of converting the received acoustic signal to a predetermined sampling rate.
  • steps (a) to (c) may be preceded by the step of extracting a signal vector from the acoustic signal, in particular, the converted acoustic signal.
  • the signal vector may be obtained by sub-sampling the acoustic signal and may comprise a predefined number of entries. Then, subsequent (in time) signal vectors may overlap. The use of signal vectors simplifies further processing of the signals.
  • Steps (a) to (c) may be preceded by the step of determining a spectral vector of the received acoustic signal.
  • a window function may be applied to signal vectors of the received acoustic signal.
  • a Hann or a Hamming window may be used (see K. D. Kammeyer, K. Kroschel, Digitale Signaltechnik, 4 th Edition, Teubner, Stuttgart, Germany 1997 ).
  • Signal vectors, in particular the signal vectors weighted in this way may be transformed into the Fourier domain using a discrete Fourier transform.
  • the resulting vector is a short-term spectral vector. This allows for further processing in the Fourier domain.
  • step (b) comprises determining a broadband spectral envelope signal and a broadband excitation signal between the lower and upper broadband bandwidth limits such that the product of spectral envelope signal and excitation signal corresponds to the received acoustic signal according to a predetermined criterion.
  • Such a decomposition into an envelope signal and an excitation signal simplifies determining the current bandwidth limits and increases the accuracy when determining a complementary signal.
  • Step (a) comprises comparing a determined broadband spectral envelope signal and a spectrum of the received acoustic signal. It turned out that the spectrum is a suitable basis for determining current bandwidth limits of the acoustic signal.
  • determining a complementary signal in step (b) based on these current bandwidth limits and comprising determination of an envelope signal enables to iteratively adapt the current bandwidth limits by comparing again the (newly) determined envelope signal and a spectrum.
  • determining current bandwidth limits in step (a) may use a spectral envelope signal determined according to step (b), particularly in a preceding step or in a preceding iteration of the method.
  • the comparing step may comprise selecting the minimal and maximal frequency for which the long-term power spectrum is larger than or equal to the determined broadband spectral envelope signal plus a predetermined constant.
  • the predetermined constant can be chosen based on empirical or theoretical data.
  • the predetermined constant may be negative.
  • determining a broadband spectral envelope signal may comprise selecting an envelope signal from a codebook according to a predetermined criterion.
  • codebooks By using codebooks, the required computing power can be reduced for determining an envelope signal.
  • different kinds of criteria can be used when selecting an envelope signal from a codebook.
  • using a predetermined distance criterion such as a cepstral distance can be used, particularly if the codebook entries have the form of cepstral vectors.
  • selecting an envelope signal may comprise equalizing the received acoustic signal and selecting an envelope signal from the codebook having minimal distance to the equalized acoustic signal according to a predetermined distance criterion, in particular, having a minimal cepstral distance.
  • Equalizing the acoustic signal allows to modify it such that a comparison with envelope signals from the codebook can be simplified.
  • the received acoustic signal can be equalized in such a way that the resulting signal shows a long-term power spectrum corresponding to the long-term power spectrum of the signal used for training the codebook.
  • Equalizing can be restricted to frequencies between the current upper and lower bandwidth limits of the received acoustic signal; outside these limits, the signal may remain unchanged.
  • equalizing the received acoustic signal can be performed using a normalized long-term power spectrum of the signal used for training the codebooks, particularly using the normalized long-term power spectrum divided by the normalized long-term power spectrum of the received acoustic signal itself.
  • the codebook may comprise pairs of corresponding envelope signals, each pair comprising a broadband envelope signal between the lower and upper broadband bandwidth limits and a corresponding narrowband envelope signal between a lower narrowband bandwidth limit being larger than the lower broadband bandwidth limit and an upper narrowband bandwidth limit being smaller than the upper broadband bandwidth limit, and selecting an envelope signal may comprise determining a narrowband envelope signal having minimal distance to the equalized acoustic signal according to the predetermined distance criterion and selecting the corresponding broadband envelope signal of this pair.
  • the received acoustic signal When using a cepstral distance to select an envelope signal, the received acoustic signal, particularly in its equalized form, has to be transformed into the cepstral domain.
  • the step of selecting an envelope signal can further comprise the steps of determining the absolute value squared of the sub-band signals of the received acoustic signal, determining an auto-correlation in the time domain, particularly by performing an inverse discrete Fourier transform on the vector of the absolute value squared, determining prediction coefficients, particularly using the Levinson-Durbin algorithm, performing a recursion to obtain the cepstral coefficients.
  • the method may further comprise the steps of recursively transforming a cepstral vector into prediction error coefficients, augmenting the prediction error filter vector by adding a predetermined number of zeros and subsequently performing a discrete Fourier transform to obtain an inverse spectrum, determining the reciprocal of each sub-band component to obtain a spectral envelope vector.
  • the step of selecting an envelope signal may be preceded by providing adapted narrowband codebook envelope signals being adapted to the current lower and upper bandwidth limits.
  • Such an adaptation of the codebook entries allows for an improved selection of a corresponding envelope signal from the codebook.
  • the adaptation would result in envelope signals in the codebook having an extended bandwidth. In this way, particularly fricatives can be more reliably detected.
  • the providing step may comprise processing broadband codebook envelope signals using a long-term power spectrum of the received acoustic signal.
  • the long-term power spectrum may be normalized; furthermore, the long-term power spectrum of the received acoustic signal may be divided by a normalized long-term power spectrum of a broadband signal used for training of the codebook.
  • the processing of the broadband codebook envelope signals may be performed only for frequencies outside the current bandwidth limits; within the bandwidth limits, the envelope signals may remain unchanged.
  • Processing using the long-term power spectrum may comprise weighting broadband codebook envelope signal vectors using the long-term power spectrum of the received acoustic signal.
  • determining a long-term power spectrum may comprise performing a first order recursive smoothing of the absolute values squared of the sub-band signals corresponding to the acoustic signal. This can be done, in particular, only if a wanted signal, such as a speech signal, has been detected in the received acoustic signal.
  • the long-term power spectrum may be normalized, particularly with respect to a long-term power spectrum within predetermined frequency limits.
  • the long-term power spectrum may be determined in the time domain. This can be done by determining the auto-correlation and performing an LPC analysis to obtain corresponding prediction coefficients.
  • determining a broadband excitation signal may be based on prediction error filtering and/or a non-linear characteristic. In this way, suitable excitation signals can be generated. Possible non-linear characteristics are disclosed, for example, in U. Kornagel, Spectral Widening of the Excitation Signal for Telephone-Band Speech Enhancement .
  • the at least one complementary signal may be based on a product of the determined broadband spectral envelope and the determined broadband excitation signal, and step (c) may comprise summing the received acoustic signal between the current lower and upper bandwidth limits and the at least one complementary signal being restricted to the band between the lower broadband bandwidth limit and a current lower bandwidth limit and/or to the band between the current upper bandwidth limit and the upper broadband bandwidth limit.
  • the complementary signal is based on spectrally coloring the excitation signal using the envelope signal.
  • Step (c) may also comprise adapting the power of the complementary signal and/or the received acoustic signal. With this step, the power of the received acoustic signal can be maintained.
  • At least one of the steps may be performed in the cepstral domain. Particularly if the entries of the codebook are cepstral vectors, this allows for performing the method in a simpler way.
  • Steps (a) to (c) of the above methods may be repeated at predetermined time intervals. Then, the repeated adaptation to the currently received acoustic signal leads to a permanent high quality of the resulting broadband signal.
  • Steps (a) to (c) of the above methods may be repeated only if a wanted signal component, such as speech activity, is detected in the received acoustic signal.
  • a wanted signal component such as speech activity
  • an extension of the bandwidth of the received acoustic signal is advantageous.
  • restricting the method to the case of detected speech activity reduces the required computing power and avoids the presence of artifacts due to mal-adaptation.
  • the invention also provides a computer program product comprising one or more computer-readable media having computer-executable instructions for performing the steps of the above-described methods when run on a computer.
  • an apparatus for providing an acoustic signal with extended bandwidth comprising:
  • such an apparatus provides an advantageous way to extend the bandwidth of a received acoustic signal.
  • the quality of the resulting output signal is increased compared to the case of bandwidth extension systems with fixed parameters.
  • the complementary signal means comprises a means for determining a broadband spectral envelope signal and a broadband excitation signal between the lower and upper broadband bandwidth limits such that the product of spectral envelope signal and excitation signal corresponds to the received acoustic signal according to a predetermined criterion.
  • the bandwidth determining means is configured to compare a determined broadband spectral envelope signal and a spectrum of the received acoustic signal.
  • the bandwidth determining means may be configured to select the minimal and maximal frequency for which the spectrum is larger than or equal to the power spectrum of the determined broadband spectral envelope signal plus a predetermined constant.
  • the means for determining a broadband spectral envelope signal may comprise a means for selecting an envelope signal from a codebook according to a predetermined criterion.
  • the means for selecting an envelope signal may be configured to equalize the received acoustic signal and select an envelope signal from the codebook having minimal distance to the equalized acoustic signal according to a predetermined distance criterion, in particular, having a minimal cepstral distance.
  • the codebook may comprise pairs of corresponding envelope signals, each pair comprising a broadband envelope signal between the lower and upper broadband bandwidth limits and a corresponding narrowband envelope signal between a lower narrowband bandwidth limit being larger than the lower broadband bandwidth limit and an upper narrowband bandwidth limit being smaller than the upper broadband bandwidth limit
  • the means for selecting an envelope signal may be configured to determine a narrowband envelope signal having minimal distance to the equalized acoustic signal according to the predetermined distance criterion and to select the corresponding broadband envelope signal of this pair.
  • the means for determining a broadband spectral envelope signal may comprise a means for providing adapted narrowband codebook envelope signals being adapted to the current lower and upper bandwidth limits.
  • the means for providing may be configured to process the broadband codebook envelope signal using a long-term power spectrum of the received acoustic signal.
  • the means for determining a broadband excitation signal may be configured to determine the broadband excitation signal based on prediction error filtering and/or a non-linear characteristic.
  • the at least one complementary signal may be based on a product of the determined broadband spectral envelope and the determined broadband excitation signal, and the assembling means may be configured to sum the received acoustic signal between the current lower and upper bandwidth limits and the at least one complementary signal being restricted to the band between the lower broadband bandwidth limit and a current lower bandwidth limit and/or to the band between the current upper bandwidth limit and the upper broadband bandwidth limit.
  • At least one of the means may be configured to perform at least part of its function in the cepstral domain.
  • the means of the above-described apparatus may be configured to perform their respective function repeatedly at predetermined time intervals.
  • the apparatus may further comprise a wanted signal detector, in particular, a speech detector, and the means may be configured to perform their respective function only if a wanted signal component is detected in the received acoustic signal.
  • a wanted signal detector in particular, a speech detector
  • Fig. 1 shows the structure of the signal flow in an apparatus for providing an acoustic signal with extended bandwidth.
  • Fig. 2 is a flow diagram illustrating an example of a method for providing an acoustic signal with extended bandwidth which could be performed by the apparatus corresponding to Fig. 1 . In view of this, Fig.'s 1 and 2 will be described in the following simultaneously.
  • an acoustic signal such as a speech signal
  • a telephone line Because of the restricted bandwidth of the telephone line, an extension of the bandwidth is desired to improve the signal quality.
  • the signal is to be augmented so as to obtain a predetermined broader bandwidth. It is to be understood that the method described in the following can be used for bandwidth extension independent of the type of incoming signal and independent of the type of transmission line, i.e., it need not be a telephone line.
  • the acoustic signal x(n) received by block 101 has already been pre-processed by increasing the sampling rate up to the predetermined broadband or wideband bandwidth. In this way, however, no additional frequency components are generated. This can be achieved, for example, by using suitable anti-aliasing or anti-imaging filters.
  • This kind of bandwidth extension preferably, is performed only for the "missing" frequency ranges; in the case of an analog telephone line, these ranges may be between 0 and 300 Hz and 3400 Hz up to half of the desired sampling rate, for example, up to 3700 Hz.
  • signal vectors x ( n ) are generated (step 202). This can be achieved by taking every r sampling values up to a certain length.
  • the elements of this matrix can be chosen corresponding to different kinds of windows. Typical windows are the Hann or Hamming window.
  • the resulting short-term spectral vector has the form: X w n [ X e j ⁇ ⁇ 0 ⁇ n , X e j ⁇ ⁇ 1 ⁇ n , ... , X e j ⁇ ⁇ ⁇ ⁇ n , ... , X ⁇ e j ⁇ ⁇ N DFT - 1 ⁇ n ⁇ ] T , wherein ⁇ ⁇ denotes the frequency variable.
  • a long-term power spectrum of the received acoustic signal is determined in block 102 (step 204).
  • 2 diring speech activity S ⁇ xx ⁇ ⁇ ⁇ , n - 1 , else .
  • the time constant ⁇ fre is chosen to be close to 1 (0 ⁇ ⁇ fre ⁇ 1) so as to obtain a sufficiently large averaging time.
  • the recursive smoothing according to the first line of the above equation may be performed continuously. However, in order to avoid any artefacts, it may be performed only if a wanted signal component is present in the received acoustic signal, for example, if speech activity is detected.
  • a speech detector may be provided as described, for example, in E. Hänsler, G. Schmidt, Acoustic Echo and Noise Control - A Practical Approach, Wiley, Hoboken, NJ, USA, 2004 .
  • the band limits ⁇ ⁇ l and ⁇ ⁇ u denote the lower and upper limits of a predefined frequency band.
  • this frequency band may correspond to a telephone band with minimal bandwidth for which the present method is to be used, for example, the limits may be 400 Hz and 3300 Hz.
  • the limits correspond to a band which is smaller or at most equal to the frequency band of the narrow frequency band within which the codebook described below has been trained; these limits being denoted by ⁇ l and ⁇ u .
  • an estimation can be performed in the time domain as well. For this purpose, an auto-correlation is estimated for about 10 to 20 sampling cycles of offset. Afterwards, prediction coefficients can be determined using an LPC (linear predictive coding) analysis.
  • LPC linear predictive coding
  • the acoustic signal is equalized.
  • ⁇ l ( n -1 ) and ⁇ u (n -1) denote the current lower and upper bandwidth limits of the received acoustic signal.
  • the bandwidth limits at time ( n -1) are taken as the current bandwidth limits.
  • S x ⁇ x ⁇ ,norm ( ⁇ ⁇ , n ) denotes the normalized long-term power spectrum of the broadband signal which has been used for training the codebook. Normalizing of such a power spectrum is performed analogously to the case of the long-term power spectrum of the received acoustic signal described above. An example for such a normalized long-term power spectrum used for training a codebook is shown in Figure 3 .
  • the acoustic signal is equalized only within the current bandwidth limits one time step before. Outside these bandwidth limits, no equalizing takes place.
  • An envelope signal corresponding to the received acoustic signal will be determined using a codebook.
  • the used codebook comprises a number of pairs of corresponding narrowband and broadband envelope signals.
  • the codebook has been obtained by training with a large database on the basis of a starting long-term power spectrum (see Y. Linde, A. Buzo, R. M. Gray, An Algorithm for Vector Quantizer Design, IEEE Trans. Comm., vol. COM-28, no. 1, pages 84 - 95, Jan. 1980 ).
  • the codebook entries are adapted in step 206 (block 104).
  • the narrowband codebook entries c i,s ( n ) are adapted.
  • the broadband envelope signals are provided as cepstral vectors c i,b ( n )
  • the corresponding spectra C i,b ( n ) are determined.
  • cepstral vectors are determined from the resulting spectral narrowband envelopes.
  • step 207 The conversion from spectral vectors to cepstral vectors and vice versa will be described in the following with respect to step 207 in which broadband spectral envelopes are determined (block 105).
  • a broadband spectral envelope from the codebook matching the acoustic signal best is determined by comparing the narrowband codebook entries with the spectral envelope of the spectrum of the acoustic signal (after equalizing).
  • the narrowband codebook entry is selected that has the smallness distance to the acoustic signal spectrum. In principle, different distance criteria can be used.
  • the cepstral distance is particularly useful as the codebook entries are provided in the form of cepstral vectors.
  • the corresponding broadband codebook entry is determined as the optimal broadband spectral envelope for the received acoustic signal. Due to the adaptation of the narrowband codebook entries as described above, an optimal narrowband envelope can be selected in a very reliable way.
  • Converting a spectral vector, particularly of the received acoustic signal, to a cepstral vector can be achieved by:
  • the optimal cepstral vector of the broadband codebook is designated by c opt,b ( n ).
  • Fig. 4 illustrates an example of a codebook with four pairs of entries.
  • a corresponding original narrowband envelope, and a corresponding adapted narrowband envelope are shown.
  • the original broadband and narrowband codebook entries have been obtained on the basis of a large database for an ISDN telephone connection.
  • the resulting optimized entries have a higher upper limit frequency. This allows for an improved detection of fricatives.
  • step 208 an excitation signal corresponding to the received acoustic signal is generated.
  • This broadband excitation signal shows a spectrally flat envelope. It corresponds to a signal which would be recorded directly behind the vocal cords.
  • the spectral envelope of the equalized short-term spectrum X eq ( n ) is estimated in the form of prediction error filter coefficients. Applying an inverse discrete Fourier transform on this spectral vector allows to determine the corresponding time signal. After that, the vector in the time domain is filtered by a prediction error filter. The corresponding filter coefficients are those that have been determined previously.
  • a non-linear characteristic such as a two-way rectification or squaring, is applied to the filtered time domain vector. This generates the missing low frequency and high frequency signal components.
  • a transformation in the Fourier domain provides, then, the spectrum of the extended excitation signal X exc ( n ) .
  • determining an excitation signal can be performed in the time sub-band or Fourier domain as well. Examples for this alternative can be found in B. Iser, G. Schmidt, Bandwidth Extension of Telephony Speech, Eurasip Newsletter, Volume 16, Number 2, pages 2 to 24, June 2005 .
  • 2 ⁇ ⁇ ⁇ l ⁇ u Y erw ( e j ⁇ ⁇ ⁇ , n ⁇ ) 2 wherein ⁇ ⁇ l and ⁇ ⁇ u denote the same bandwidth limits as in the estimation of the long-term power spectrum above.
  • the current bandwidth limits are adapted in step 210 (block 108).
  • 2 ⁇ C opt , b ⁇ e j ⁇ ⁇ ⁇ , n 2 + K C , ⁇ u n min ⁇ ⁇ ⁇
  • Fig. 5 an example for determining the bandwidth limits is illustrated.
  • the above, intermediate limit values are given by the points of intersection between the lowered broadband spectral envelope and the spectrum of the received acoustic signal.
  • These intermediate limit values may be recursively smoothed to eliminate temporary mal-estimations.
  • smoothing is performed only if speech activity is detected in the current signal frame.
  • the received acoustic signal is passed through an adaptive band pass filter to retain only components within the current bandwidth limits (block 109) to obtain a spectral vector Y tel ( n ).
  • the spectrally colored excitation signal is passed through a complementary adaptive band stop filter (block 110) so as to obtain a vector Y ext ( n ).
  • Y tel n G tel n ⁇ X w n
  • Y ext n G ext n ⁇ X ext n
  • the weighting matrices G tel ( n ) and G ext ( n ) are diagonal matrices:
  • G tel n G tel e j ⁇ ⁇ 0 ⁇ n 0 ... 0 0
  • G ext n G ext e j ⁇ ⁇ 0 ⁇ n 0 ... 0 0
  • the transitions at the bandwidth limits can be realized in a smoother way.
  • the resulting time domain vectors are, then, assembled using an overlap add method (as described in K. D. Kammeyer, K. Kroschel, Digitale Signalmaschine ) to obtain the final output signal y ( n ).
  • the steps performed in the Fourier domain may also be performed in the time domain.
  • equalizing the acoustic signal may be performed when adapting the narrowband codebook entries.
  • the above-described equalizing step may be augmented. For example, if an amplification or an attenuation is detected at particular frequencies, it may be adjusted within the bandwidth limits as well. In this case, the output vector Y tel ( n ) is modified with the weighting matrix H mod ( n ).

Claims (23)

  1. Procédé pour mettre à disposition un signal acoustique avec une largeur de bande étendue, comprenant :
    a) la détermination automatique d'une limite de largeur de bande courante plus basse et plus haute d'un signal acoustique reçu,
    b) la détermination automatique d'au moins un signal complémentaire pour compléter le signal acoustique reçu entre une limite prédéterminée de largeur de bande à bande large plus basse et la limite de largeur de bande courante plus basse et/ou entre la limite de largeur de bande courante plus haute et une limite prédéterminée de largeur de bande à bande large plus haute, la limite prédéterminée de largeur de bande à bande large plus basse étant plus petite que la limite de largeur de bande courante basse et la limite prédéterminée de largeur de bande à bande large plus haute étant plus grande que la limite de largeur de bande actuelle plus haute,
    c) l'assemblage automatique du, au moins un, signal complémentaire et du signal acoustique reçu pour obtenir un signal acoustique avec une largeur de bande étendue,
    dans lequel l'étape (b) comprend la détermination d'un signal d'enveloppe spectrale à bande large et un signal d'excitation à bande large entre les limites de largeur de bande plus haute et plus basse tel que le produit du signal d'enveloppe spectrale et le signal d'excitation correspond au signal acoustique reçu selon un critère prédéterminé, et
    dans lequel l'étape (a) comprend la comparaison du signal d'enveloppe spectrale à bande large déterminé et d'un spectre du signal acoustique reçu telle que la limite de largeur de bande courante plus haute et la limite de largeur de bande courante plus basse soient itérativement adaptées, et où l'étape de comparaison comprend la sélection de la fréquence minimale et maximale pour lesquelles le spectre est plus grand ou égal au signal d'enveloppe spectrale à bande large déterminé ajouté à une constante prédéterminée.
  2. Procédé selon la revendication 1, dans lequel la détermination d'un signal d'enveloppe spectrale à bande large comprend la sélection d'un signal d'enveloppe d'un livre code selon un critère prédéterminé.
  3. Procédé selon la revendication 2, où la sélection d'un signal d'enveloppe comprend l'égalisation du signal acoustique reçu et la sélection d'un signal d'enveloppe à partir du livre code ayant une distance minimale du signal acoustique égalisé selon un critère distance prédéterminé, en particulier ayant une distance cepstral minimale.
  4. Procédé selon la revendication 3, où le livre code comprend des paires de signaux d'enveloppe correspondants, chaque paire comprenant un signal d'enveloppe à bande large entre les limites de largeur de bande à bande large plus basse et plus haute et un signal d'enveloppe à bande étroite correspondant entre une limite de largeur de bande à bande basse plus basse étant plus large que la limite de largeur de bande à bande large basse et une limite de largeur de bande à bande base plus haute étant plus petite que la limite de largeur de bande à bande large plus haute, et la sélection d'un signal d'enveloppe comprend la détermination d'un signal d'enveloppe à bande basse ayant une distance minimale du signal acoustique égalisé selon le critère de distance prédéterminée et la sélection du signal d'enveloppe à bande large correspondant de cette paire.
  5. Procédé selon la revendication 4, dans lequel l'étape de sélection d'un signal d'enveloppe est anticipée par la mise à disposition de signaux d'enveloppe de livre code à bande étroite adaptés étant adaptés aux limites de largeur de bande plus basse et plus haute courantes.
  6. Procédé selon la revendication 5, dans lequel l'étape de mise à disposition comprend le traitement des signaux d'enveloppe de livre code à bande large en utilisant un spectre de puissance à long terme du signal acoustique reçu.
  7. Procédé selon une des revendications 1 à 6, dans lequel la détermination d'un signal d'excitation à bande large est basé sur un filtrage d'erreur de prédiction et/ou une caractéristique non linéaire.
  8. Procédé selon une des revendications 1 à 7, dans lequel le, au moins un, signal complémentaire est basé sur le produit de l'enveloppe spectrale à bande large déterminée et le signal d'excitation à bande large déterminé, et où l'étape (c) comprend la sommation du signal acoustique reçu entre les limites de largeur de bande plus basse et plus haute courantes et le, au moins un, signal complémentaire étant restreint à la bande entre la limite de largeur de bande à bande large plus basse et la limite de largeur de bande plus basse courante et/ou à la bande entre la limite de largeur de bande plus haute courante et la limite de largeur de bande à bande large plus haute.
  9. Procédé selon une des revendications précédentes, dans lequel au moins une des étapes est réalisée dans le domaine cepstral.
  10. Procédé selon une des revendications précédentes, dans lequel les étapes (a) à (c) sont répétées à des intervalles temporels prédéterminés.
  11. Procédé selon une des revendications précédentes, dans lequel les étapes (a) à (c) sont répétées seulement si un composant d'un signal désiré, en particulier activité de parole, est détecté dans le signal acoustique reçu.
  12. Produit de programme d'ordinateur comprenant un ou plusieurs médias lisible par ordinateur ayant des instructions pouvant être exécutées sur ordinateur pour réaliser les étapes du procédé selon une des revendications précédentes lorsqu'elles sont réalisées sur ordinateur.
  13. Appareil pour mettre à disposition un signal acoustique avec une largeur de bande étendue, comprenant :
    un moyen de détermination de largeur de bande pour déterminer automatiquement une limite de largeur de bande actuelle plus haute et plus basse d'un signal acoustique reçu,
    un moyen de signal complémentaire pour déterminer automatiquement au moins un signal complémentaire pour compléter le signal acoustique reçu entre une limite prédéterminée de largeur de bande à bande large plus basse et la limite de largeur de bande courante plus basse et/ou entre la limite de largeur de bande courante plus haute et une limite prédéterminée de largeur de bande à bande large plus haute, ou la limite prédéterminée de largeur de bande à bande large plus basse est plus petite que la limite de largeur de bande courante plus basse et la limite prédéterminée de largeur de bande à bande large plus haute est plus grande que la limite de largeur de bande courante plus haute, et
    un moyen d'assemblage pour l'assemblage automatique du, au moins un, signal complémentaire et du signal acoustique reçu pour obtenir un signal acoustique avec une largeur de bande étendue,
    dans lequel le moyen de signal complémentaire comprend un moyen pour déterminer un signal d'enveloppe spectrale à bande large et un signal d'excitation à bande large entre les limites de largeur de bande plus haute et plus basse telle que le produit du signal d'enveloppe spectrale et le signal d'excitation correspond au signal acoustique reçu selon un critère prédéterminé, et
    dans lequel le le moyen de détermination de largeur de bande est configuré pour comparer le signal d'enveloppe spectrale à bande large déterminé et un spectre du signal acoustique reçu telle que la limite de largeur de bande courante plus haute et la limite de largeur de bande courante basse soient itérativement adaptées, et où le moyen de détermination de largeur de bande est configuré pour sélectionner la fréquence minimale et maximale pour lesquelles le spectre est plus grand ou égal au signal d'enveloppe spectrale à bande large déterminé plus une constante prédéterminée.
  14. Appareil selon la revendication 13, dans lequel le moyen de détermination d'un signal d'enveloppe spectrale à bande large comprend un moyen pour sélectionner un signal d'enveloppe d'un livre code selon un critère prédéterminé.
  15. Appareil selon la revendication 14, dans lequel le moyen pour sélectionner un signal d'enveloppe est configuré pour égaliser le signal acoustique reçu et pour sélectionner un signal d'enveloppe à partir du livre code ayant une distance minimale du signal acoustique égalisé selon un critère distance prédéterminé, en particulier ayant une distance cepstrale minimale.
  16. Appareil selon la revendication 15, où le livre code comprend des paires de signaux d'enveloppe correspondants, chaque paire comprenant un signal d'enveloppe à bande large entre les limites de largeur de bande à bande largue plus basse et plus haute et un signal d'enveloppe à bande étroite correspondant entre une limite de largeur de bande à bande basse plus basse étant plus large que la limite de largeur de bande à bande large basse et une limite de largeur de bande à bande base plus haute étant plus petite que la limite de largeur de bande à bande large plus haute, et le moyen de sélection d'un signal d'enveloppe est configuré pour déterminer un signal d'enveloppe à bande basse ayant une distance minimale du signal acoustique égalisé selon le critère de distance prédéterminé et pour sélectionner le signal d'enveloppe à bande large correspondant de cette paire.
  17. Appareil selon la revendication 16, dans lequel le moyen pour déterminer un signal d'enveloppe à bande large comprend un moyen pour mettre à disposition des signaux d'enveloppe de livre code à bande étroite adaptés étant adaptés aux limites de largeur de bande plus basse et plus haute actuelles.
  18. Appareil selon la revendication 17, dans lequel le moyen de mise à disposition est configuré pour traiter le signal d'enveloppe de livre code à bande large en utilisant un spectre de puissance à long terme du signal acoustique reçu.
  19. Appareil selon une des revendications 13 à 18, dans lequel le moyen pour déterminer un signal d'excitation à bande large est configuré pour déterminer le signal d'excitation à bande large basé sur un filtrage d'erreur de prédiction et/ou une caractéristique non linéaire.
  20. Appareil selon une des revendications 13 à 19, dans lequel le, au moins un, signal complémentaire est basé sur le produit de l'enveloppe spectrale à bande large déterminée et le signal d'excitation à bande large déterminé, et où le moyen d'assemblage est configuré pour sommer le signal acoustique reçu entre les limites de largeur de bande plus basse et plus haute courantes et le, au moins un, signal complémentaire étant restreint à la bande entre la limite de largeur de bande à bande large plus basse et la limite de largeur de bande plus basse actuelle et/ou à la bande entre la limite de largeur de bande plus haute actuelle et la limite de largeur de bande à bande large plus haute.
  21. Appareil selon une des revendications 13 à 20, dans lequel au moins un des moyens est configuré pour réaliser au moins une partie de sa fonction dans le domaine cepstral.
  22. Appareil selon une des revendications 13 à 21, où les moyens sont configurés pour réaliser leur fonction respective d'une façon répétitive à des intervalles temporels prédéterminés.
  23. Appareil selon une des revendications 13 à 22, comprenant en outre un capteur de signal désiré, en particulier un détecteur de parole, et où les moyens sont configurés pour réaliser leur fonction respective seulement si un composant d'un signal désiré est détecté dans le signal acoustique reçu.
EP06017456A 2006-08-22 2006-08-22 Procédé et système fournissant un signal acoustique avec une largeur de bande étendue Not-in-force EP1892703B1 (fr)

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AT06017456T ATE446572T1 (de) 2006-08-22 2006-08-22 Verfahren und system zur bereitstellung eines tonsignals mit erweiterter bandbreite
DE602006009927T DE602006009927D1 (de) 2006-08-22 2006-08-22 Verfahren und System zur Bereitstellung eines Tonsignals mit erweiterter Bandbreite
EP06017456A EP1892703B1 (fr) 2006-08-22 2006-08-22 Procédé et système fournissant un signal acoustique avec une largeur de bande étendue
CA002596411A CA2596411A1 (fr) 2006-08-22 2007-08-08 Methode et systeme de fourniture d'un signal acoustique avec largeur de bande etendue
JP2007214930A JP5150165B2 (ja) 2006-08-22 2007-08-21 拡張された帯域幅を有する音響信号を提供するための方法およびシステム
CN2007101466102A CN101141533B (zh) 2006-08-22 2007-08-22 用于提供具有扩展带宽的声音信号的方法和系统
KR1020070084306A KR101433833B1 (ko) 2006-08-22 2007-08-22 음향 신호에 확장된 대역폭을 제공하기 위한 방법 및시스템

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US8463599B2 (en) 2009-02-04 2013-06-11 Motorola Mobility Llc Bandwidth extension method and apparatus for a modified discrete cosine transform audio coder
US8688441B2 (en) 2007-11-29 2014-04-01 Motorola Mobility Llc Method and apparatus to facilitate provision and use of an energy value to determine a spectral envelope shape for out-of-signal bandwidth content

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US8433582B2 (en) 2008-02-01 2013-04-30 Motorola Mobility Llc Method and apparatus for estimating high-band energy in a bandwidth extension system
US8463599B2 (en) 2009-02-04 2013-06-11 Motorola Mobility Llc Bandwidth extension method and apparatus for a modified discrete cosine transform audio coder

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ATE446572T1 (de) 2009-11-15
JP5150165B2 (ja) 2013-02-20
CN101141533B (zh) 2013-09-04
KR101433833B1 (ko) 2014-08-27
DE602006009927D1 (de) 2009-12-03
CN101141533A (zh) 2008-03-12
EP1892703A1 (fr) 2008-02-27
JP2008052277A (ja) 2008-03-06
CA2596411A1 (fr) 2008-02-22

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