Classifications

• • G—PHYSICS
  • G10—MUSICAL INSTRUMENTS; ACOUSTICS
  • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
  • G10L21/00—Speech or voice signal processing techniques to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility
  • G10L21/02—Speech enhancement, e.g. noise reduction or echo cancellation
  • G10L21/0316—Speech enhancement, e.g. noise reduction or echo cancellation by changing the amplitude
  • G10L21/0364—Speech enhancement, e.g. noise reduction or echo cancellation by changing the amplitude for improving intelligibility
• • 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
  • G10L25/00—Speech or voice analysis techniques not restricted to a single one of groups G10L15/00 - G10L21/00
  • G10L25/03—Speech or voice analysis techniques not restricted to a single one of groups G10L15/00 - G10L21/00 characterised by the type of extracted parameters
• • 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
  • G10L25/00—Speech or voice analysis techniques not restricted to a single one of groups G10L15/00 - G10L21/00
  • G10L25/03—Speech or voice analysis techniques not restricted to a single one of groups G10L15/00 - G10L21/00 characterised by the type of extracted parameters
  • G10L25/18—Speech or voice analysis techniques not restricted to a single one of groups G10L15/00 - G10L21/00 characterised by the type of extracted parameters the extracted parameters being spectral information of each sub-band

Definitions

• the invention is a method for improving the speech quality of an electro-laryngeal (EL) speaker, wherein the speech signal of the speaker is digitized by suitable means.
• suitable means is meant, for example, a microphone with associated analog / digital converter, a telephone or other methods using electronic equipment.
• An EL is an artificial spare voice device, for example, for patients who have had their larynx surgically removed.
• the EL is attached to the underside of the jaw; a tone generator with a certain frequency makes the air in the oral cavity vibrate over the soft tissues on the underside of the jaw. These vibrations are then modulated by the articulation organs so that speaking becomes possible.
• the tone generator usually only works with one frequency, the voice sounds monotonous and unnatural, or "robotic".
• the vibration of the EL disturbs the perception of the speech or even drowned out, because only a part of the sound is articulated in the oral cavity.
• the parts emerging directly from the device or at the transition point on the neck overlay the articulated parts and reduce the intelligibility. This is particularly the case with speakers who have received radiation therapy in the neck area, which stiffens the tissue structure.
• Various methods have therefore been developed which are intended to amplify the useful signal - ie the articulated vibrations - in relation to the interference signal - ie the direct sound or the unmodulated vibration of the EL.
• this algorithm adapts the subtraction parameters in the frequency domain based on auditory masking. It is assumed that speech and background noise are uncorrelated and therefore the background noise can be estimated and subtracted from the signal in the frequency domain.
• US 2005/0004604 A1 describes a laryngeal solution in which a sounder and a microphone are placed directly in front of the mouth of a user, wherein the sounder emits a sound at low volume and the signal for further processing via the microphone is recorded.
• the signal is essentially filtered with a comb filter to reduce or remove the harmonics of the signal.
• the quality of the speech signal is badly affected.
• WO 2006/099670 A1 describes a device for monitoring the respiratory tract, wherein sound in the audible frequency range is introduced into the respiratory tract of an object and the condition of the respiratory tract is determined from the reflected or processed sound. For example, it is possible to detect airway obstruction.
• the exceeding of specific threshold values is checked by means of the FFT (fast Fourier transformation), from which conclusions are drawn regarding the treatment of the measured signal.
• This object is achieved by a method of the type mentioned according to the invention by the following steps: a) dividing a single-channel speech signal into a series of frequency channels by transferring the time domain into a discrete frequency range, b) filtering out the modulation frequency of the EL by means of a high-pass or Notch filter in each frequency channel, and c) retransforming the filtered speech signal from the frequency domain to the time domain and merging into a single channel output signal.
• the invention makes use of an improved model of the use of an EL, according to which the EL fundamental sound articulated to a speech signal as well as the unaltered portions of the EL interfering with the perception of the speech signal come from a common source, namely the EL.
• the person skilled in the art knows a large number of possibilities for converting a digitized, single-channel signal into the frequency domain and thus dividing it into a series of frequency channels.
• the modulation frequency of the EL is filtered by suitable filters - e.g. Notch or high pass filter applied to the amount - suppresses and thus improves the quality of the articulated signal components.
• the method according to the invention thus aims to increase the intelligibility of the language of EL users or to make the signal more pleasant and "more human.”
• the aim is to reduce the direct sound from the EL during communication via electronic means (eg telephone) or to eliminate.
• the implementation of the method according to the invention can be done for example by a software plug-in, as a hard-wired solution or as an analog circuit.
• the conversion in step a) of the method according to the invention is advantageously carried out by Fourier transformation and the inverse transformation in step c) by means of inverse Fourier transformation.
• the transfer takes place in blocks (eg blocks of 20 ms) at short intervals (refresh every 10 ms, for example).
• the division of the signal into a series of frequency channels takes place when transferring the signal into the frequency domain.
• the transfer of the speech signal in step a) and the inverse transformation in step c) with a corresponding filter bank is advantageously carried out by Fourier transformation and the inverse transformation in step c) by means of inverse Fourier transformation.
• the results of the method according to the invention can be further improved if signal compression takes place before the filtering in step b) and decompression takes place after step b).
• the compression can be prevented that at high amplitudes whose changes are so dominant that the changes of small amplitudes are not taken into account. Compression makes relative changes more visible to the filter.
• step c before the inverse transformation in step c), a rectification of the negative signal components takes place.
• Fig. 2 is a simplified representation of the situation in which the method according to the invention finds application and
• Fig. 3 is a block diagram of the method according to the invention.
• Fig. 1 the different transmission paths of the signal of an EL 1 are outlined.
• an EL 1 is arranged on the neck of a speaker 2.
• the sound produced by the EL 1 spreads on the one hand through the normal speech channels (mouth and nose) 5 of the first speaker 2 and is there articulated to language; this first signal 3 is clearly variable, or time-variant.
• a second signal 6 shown in phantom in Fig. 1 in the form of direct sound of EL 1, this signal 4 is largely stationary and is therefore assumed to be invariant in time.
• the second part 6 of the overall signal that is the background noise of the EL 1
• the second part 6 of the overall signal is perceived by the listener 4 as an interference signal and reduces the intelligibility of the speech of the speaker 2.
• the original excitation by means of the EL 1 is thus transmitted via two different paths.
• the invention relates to the improvement of the speech quality of an EL speaker when using electronic mediators - so instead of a listener, the signals would be recorded, for example, with a microphone. To illustrate the However, for the sake of clarity, this general model has been chosen.
• FIG. 2 shows a simplified model representation of the situation to which the inventive method for suppressing a disturbing second signal 6 (see FIG. 1) is applied. It is readily apparent that in the method according to the invention there is no separation of signal sources, but of propagation paths.
• a source signal x (w) from a signal source 7 propagates over two different signal paths.
• the output signal is modulated by a time-variant filter H (w, t) to a time-variant signal x (w) H (w, t).
• the output signal is changed only by a time-invariant filter F (w) to a signal x (w) F (w).
• the signals of the two paths are then received in a receiver 8 - e.g. the ear of a listener, a microphone o.a. - Summed to a signal available for measurement S (w, t).
• the inarticulate signal component x (w) F (w) (ie the background noise of the EL) superimposes the time-variant speech signal x (w) H (w, t) and thereby causes a loss of intelligibility for the speech signal.
• Speech intelligibility is improved by separating the time-variant signal component from the time-invariant signal component.
• Fig. 3 shows a possible implementation of the method according to the invention.
• an arbitrary digital speech signal 9 from a speaker with EL can be present at the input.
• the speech signal 9 is transformed in blocks into the frequency domain and thus divided into a series of frequency channels.
• the person skilled in the art can choose here from various established methods for the transformation of a signal from the time domain into the frequency domain;
• the discrete cosine transformation is also used - however, the prerequisite for an application according to the invention is that the transformation is reversible.
• the signal is divided at a certain refresh rate (eg 10 ms) into blocks of, for example, 20 ms in length, which are each fanned out into a series of frequency channels 11.
• the originally single-channel speech signal 9 is thus split into a plurality of frequency ranges, which change as a result of time.
• the frequency signal is complex, but subsequently only the absolute value is modified, phase 15 remains unchanged.
• a filter bank may also be used, reducing the sample rate of the signal after the filter bank.
• the reduction of the sampling rate corresponds to the block formation when using the Fourier transform.
• each frequency channel 11 is filtered, for example with a high-pass filter or notch filter.
• This filtering allows certain frequencies to be filtered out - in audio engineering, narrow-band interferences are eliminated with notch filters. Since the EL oscillates at a certain frequency - for example 100 Hz - the interference signal, which is not changed by the articulation organs of a speaker, results in the frequency range amplitudes in the 100 Hz channel with the modulation frequency 0 Hz - ie the amplitude of the EL Signal does not change.
• the interference signal is characterized in that it is perfectly time-invariant.
• a notch or a high pass filter are used to filter the background noise of the EL.
• the limiting frequency for the high-pass filter is the modulation frequency of the EL; the notch filter is chosen so that it locks exactly at the modulation frequency of the EL.
• the filter is not restricted to just one frequency but covers a specific frequency range-in this case a modulation frequency range-the function of the method according to the invention is ensured.
• a final function block 13 the feedback of the signals into the time domain, for example by means of inverse Fourier transformation and the merger of the frequency channels 11 back into a channel by means of overlap-add.
• the overlap-add method is a method of digital signal processing known to the person skilled in the art.
• the result is a single-channel output signal 14, in which the interference signal of the EL is filtered out or at least attenuated.
• the output signal can then be processed further.
• the sampling rate of the signal is increased again after the filtering in step 12 and then treated further as described.
• the invention can be used, for example, as an additional device for telephoning.
• the device In a conventional analogue telephone, the device is simply integrated into the handset.
• the integration of the invention by a software plug-in is possible.
• the realization in the context of a hardwired solution, e.g. also in an analog circuit, is possible.
• the method according to the invention can also be used when using an EL in which two or more frequencies can be switched back and forth in order to give the speech a more realistic sound. This applies both to discrete frequency jumps and to continuous changes in the fundamental frequency assuming that the frequencies being switched are within a frequency band into which the fundamental signal is split.
• the width of the modulation frequency filter determines how fast the frequency may change. With very slow, continuous changes, the frequency can change over the entire band of the frequency band if the suppression function works - the decisive factor is not the size but the speed of the change. When switching the EL on and off, which corresponds to rapid changes, the suppression only takes a few milliseconds - depending on how wide the notch filter is selected or where the fundamental frequency of the high-pass filter is.
• the changes in the fundamental frequency must not be too large.
• the frequency channels into which the signal is split would have to be expanded, or the filtering by means of a high-pass filter would have to start at a somewhat higher frequency.

Landscapes

• Engineering & Computer Science (AREA)
• Human Computer Interaction (AREA)
• Quality & Reliability (AREA)
• Signal Processing (AREA)
• Health & Medical Sciences (AREA)
• Audiology, Speech & Language Pathology (AREA)
• Computational Linguistics (AREA)
• Physics & Mathematics (AREA)
• Acoustics & Sound (AREA)
• Multimedia (AREA)
• Soundproofing, Sound Blocking, And Sound Damping (AREA)
• Telephone Function (AREA)
• Prostheses (AREA)

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Citations (4)

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• 2009
  • 2009-02-04 AT AT0019309A patent/AT507844B1/de not_active IP Right Cessation
• 2010
  • 2010-02-01 EP EP10708882.5A patent/EP2394271B1/de not_active Not-in-force
  • 2010-02-01 JP JP2011548504A patent/JP5249431B2/ja not_active Expired - Fee Related
  • 2010-02-01 ES ES10708882.5T patent/ES2628521T3/es active Active
  • 2010-02-01 US US13/147,893 patent/US20120004906A1/en not_active Abandoned
  • 2010-02-01 CN CN201080010113.XA patent/CN102341853B/zh not_active Expired - Fee Related
  • 2010-02-01 WO PCT/AT2010/000032 patent/WO2010088709A1/de not_active Ceased
  • 2010-02-01 PT PT107088825T patent/PT2394271T/pt unknown
  • 2010-02-01 CA CA2749617A patent/CA2749617C/en not_active Expired - Fee Related
  • 2010-02-01 DK DK10708882.5T patent/DK2394271T3/en active

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