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WO2001018794A1 - Accentuation spectrale de signaux acoustiques garantissant une meilleure reconnaissance de la parole - Google Patents

Accentuation spectrale de signaux acoustiques garantissant une meilleure reconnaissance de la parole Download PDF

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Publication number
WO2001018794A1
WO2001018794A1 PCT/US2000/040854 US0040854W WO0118794A1 WO 2001018794 A1 WO2001018794 A1 WO 2001018794A1 US 0040854 W US0040854 W US 0040854W WO 0118794 A1 WO0118794 A1 WO 0118794A1
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WIPO (PCT)
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channel
signal
history
energy
digital
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PCT/US2000/040854
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English (en)
Inventor
Keith R. Kluender
Rick Lynn Jenison
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Wisconsin Alumni Research Foundation
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Wisconsin Alumni Research Foundation
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Filing date
Publication date
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Priority to AU12508/01A priority Critical patent/AU1250801A/en
Publication of WO2001018794A1 publication Critical patent/WO2001018794A1/fr
Anticipated expiration legal-status Critical
Ceased legal-status Critical Current

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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
    • G10L15/00—Speech recognition
    • G10L15/02—Feature extraction for speech recognition; Selection of recognition unit
    • 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
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R25/00—Deaf-aid sets, i.e. electro-acoustic or electro-mechanical hearing aids; Electric tinnitus maskers providing an auditory perception
    • H04R25/50—Customised settings for obtaining desired overall acoustical characteristics
    • H04R25/505—Customised settings for obtaining desired overall acoustical characteristics using digital signal processing
    • 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
    • G10L21/04—Time compression or expansion
    • G10L21/057—Time compression or expansion for improving intelligibility
    • G10L2021/0575—Aids for the handicapped in speaking
    • 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
    • G10L21/0232—Processing in the frequency domain
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R2225/00—Details of deaf aids covered by H04R25/00, not provided for in any of its subgroups
    • H04R2225/43—Signal processing in hearing aids to enhance the speech intelligibility
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R25/00—Deaf-aid sets, i.e. electro-acoustic or electro-mechanical hearing aids; Electric tinnitus maskers providing an auditory perception
    • H04R25/35—Deaf-aid sets, i.e. electro-acoustic or electro-mechanical hearing aids; Electric tinnitus maskers providing an auditory perception using translation techniques
    • H04R25/356—Amplitude, e.g. amplitude shift or compression

Definitions

  • This invention pertains generally to the field of audio signal processing and particularly to hearing aids and speech recognition.
  • Direct Realism is a general theory for all senses holding that perception is an act by which properties of the physical world that are significant to a perceiver, "distal events, " are directly recovered without intermediate construction.
  • distal events are held to be linguistically relevant articulations of the vocal tract.
  • the most critical concern with regard to this approach is that one must be able to solve the "inverse problem. "
  • the perceiver In order to recover a unique distal event in any modality, the perceiver has only the physical energy available to sensory receptors. Independent of classic concerns regarding the extent to which one should view this source of information as rich or impoverished, what must be true is that there is sufficient information to successfully make the inverse transformation to a unique distal event.
  • Cochlear hearing impairment is associated with reduced frequency selectivity and with loudness recruitment. These two factors are not independent. Elevated thresholds for hearing impaired listeners result in limited dynamic range. Once amplification has been introduced to make the signal suprathreshold, the system is in a compressive state, leading to "spectral smearing. " The consequences of this deficiency in spectral definition seem to be more severe for some aspects of the speech signal than for others. As might be expected, for example, amplitude envelope shapes suffer least when audibility is improved with amplification, probably owing to the ability to encode such information in temporal firing patterns irrespective of spectral detail. By contrast, most types of spectral information are perceived poorly even when audibility is provided. Additional amplification not only does not help in these cases, but additional increments in amplification can even lead to decreased speech recognition.
  • the present invention provides a method and apparatus for enhancing an auditory signal.
  • the present invention employs a process which enhances spectral differences between sounds in a fashion mimicking that of human auditory systems. Implementation imitates neuroprocesses of adaptation, suppression, adaptation of suppression, and descending inhibitory pathways.
  • the present invention serves to make sounds, particularly speech sounds, more distinguishable.
  • an input auditory signal is divided into a plurality of spectral channels. This may be accomplished, for example, by applying the input auditory signal to a bank of gammatone or
  • An output gain for each channel is derived based on the time varying history of energy in the channel.
  • the magnitude of the output gain thus derived is preferably inversely related to the history of energy in the channel.
  • the output gain may be derived by determining a weighted energy history of the channel, converting the weighted energy history into an RMS history weighting value, and subtracting the RMS history weighting value from unity to determine the output gain for the channel.
  • the output gain for each channel preferably also takes into consideration the time varying history of energy in neighboring spectral channels.
  • the output gain for each channel may preferably be derived by subtracting the ratio of the RMS history weighting value for the channel to a sum of RMS history weighting values for neighboring channels from unity to determine the output gain for the channel.
  • the output gain thus derived is applied to the channel to form a plurality of modified spectral channel signals.
  • the plurality of modified spectral channel signals are combined to form an enhanced output auditory signal.
  • the present invention is particularly applicable to use in electronic hearing aid devices for use by the hearing impaired, particularly for purposes of enhancing the spectrum such that impaired biological signal processing in the auditory brain stem is restored.
  • An electronic hearing aid device incorporating the present invention may include a microphone for receiving sound and converting it into electrical signals, appropriate amplification and filtering, an analog to digital converter, a signal processor, such as a digital signal processor, implementing signal processing for enhancing the auditory signal in accordance with the present invention, a digital to analog converter, output side filters and amplifiers, and a speaker for providing the enhanced auditory signal to a wearer of the hearing aid device.
  • the present invention may be employed in any system wherein it is desired to make sounds, particularly speech sounds, more distinguishable.
  • the present invention may be incorporated into a computer speech recognition system.
  • a computer speech recognition system may include a microphone that converts a sound to an analog signal presented to an amplifier and filter, the output of which is provided to an analog to digital converter, which provides digital data to a signal processor, wherein processing in accordance with the present invention to enhance the auditory signal as provided.
  • recorded signal data may be provided from a recording system directly to the signal processor.
  • the output of the signal processor is provided to a speech recognition system, which itself may be implemented in a general purpose computer, with the output of the speech recognition system provided to output devices or to digital storage media.
  • Fig. 1 is a schematic block diagram of an electronic hearing aid device incorporating a signal processor for enhancing an auditory signal in accordance with the present invention.
  • Fig. 2 is a schematic block diagram of a speech recognition system incorporating a signal processor for enhancing an auditory signal in accordance with the present invention.
  • Fig. 3 is a schematic illustration of an exemplary system for enhancing an auditory signal in accordance with the present invention.
  • the present invention provides a method and apparatus for enhancing an auditory signal.
  • the present invention may be employed in electronic hearing aid devices for use by the hearing impaired, particularly for purposes of enhancing the spectrum such that impaired biological signal processing in the auditory brainstem is restored. This process enhances spectral differences between sounds in a fashion mimicking that of non-pathological human auditory systems.
  • Implementation imitates neural processes of adaptation, suppression, adaptation of suppression, and descending inhibitory pathways.
  • the invention serves to make sounds, particularly speech sounds, more distinguishable.
  • the present invention is applicable to uses other than hearing aids, such as computer speech recognition systems.
  • the present invention is directed to solve the problem that, for many hearing-impaired listeners, amplification is required to make a signal audible, but because of limited dynamic range, spectral resolution deteriorates at amplified presentation levels. All this is assumed to take place at the cochlea.
  • the invention addresses this problem by manipulation of the spectral composition of the signal to overcome some of the loss of spectral resolution, and to substitute to some extent for additional amplification (which becomes deleterious at higher levels.)
  • the presumed locus for contrast effects is at a level substantially beyond the cochlea, and an auditory signal is appropriately modified in a manner that permits these contrast mechanisms to increase perceptual distinctiveness more centrally.
  • contrast mechanisms are intact, and, if the speech spectrum is enhanced in a manner that at least partially circumvents the limited spectral resolution of the impaired periphery, these more central processes can better distinguish speech sounds.
  • a formant that ends with closure silence and begins again (after closure) at a slightly higher or lower frequency.
  • the impaired ear there would be no perceived difference in the offset and onset frequencies, as both would be consumed within the same broadened (smeared) frequency channel. Such would not be the case for the non-impaired ear. Instead, contrastive process would serve to "repel" these spectral prominences making them more distinct.
  • a general hearing aid system 10 includes a microphone 11 for receiving sound and converting it into electrical signals, appropriate amplification and filtering 12, an analog to digital converter 13, a signal processor, such as a digital signal processor 14, which carries out the signal processing in accordance with the invention as described further below, a digital-to- analog converter 15, filter and amplifiers 16, and a speaker 17 which converts the amplified signal to sound for the hearing impaired listener.
  • the speech recognition system 20 may receive sound from a microphone 21 that converts the sound to an analog signal presented to an amplifier and filter 22, the output of which is provided to an analog to digital converter 23, which provides digital data to a signal processor, such as a digital signal processor 24, which may be implemented in a general purpose computer.
  • a signal processor such as a digital signal processor 24, which may be implemented in a general purpose computer.
  • recorded signal data may be provided from a recording system 25 directly to the signal processor 24.
  • the output of the signal processor 24 is provided to a speech recognition system 26, which itself may be a general purpose computer (and the speech recognition system 26 and the signal processor 24 may both be implemented using the same computer), with the output of the speech recognition system 26 provided to output devices 27 (hard copy, video displays, etc.), or to digital storage media 28.
  • the hardware for such systems 10 and 20 is of conventional design and is well known, with present invention being implemented in a signal processor 14 and 24.
  • adaptive filters may be utilized.
  • the essence of this approach is to shape the spectrum based upon preceding energy that passes through a bank of filters.
  • Frequencies present at time t are attenuated modestly at time t+ 1, akin to simple adaptation.
  • the filter successively attenuates energy at lower frequencies through which the spectral prominence has already passed. This will have two consequences. First, the shoulder on the low-frequency side of the formant will be sharpened because that is where most energy was immediately prior. This will serve to "sharpen" the spectrum as compensation for smearing in the impaired ear.
  • the effective frequency (center of gravity) of the formant peak will be skewed away from where the formant had been before. The consequence is that contrast will be imposed on the signal (spreading successive formants apart in frequency) . It also is the case that a formant transition will be "accelerated” via this process. Because the filter successively attenuates the low-frequency shoulder, the effective slope of the processed formant steepens. At first, one may suspect that more radical formant transitions cannot be a good thing for hearing-impaired listeners.
  • a bank of gammatone filters is constructed. Because filter skirts overlap, this method includes biorthogonal reconstruction when filter outputs are recombined.
  • a second strategy uses a large bank of filters based on perfectly reconstructing Quadrature Mirror Filters.
  • the QMF filterbank affords arbitrarily fine analysis in the low-frequency region of the spectrum akin to that afforded by the cochlea.
  • the output gain from this first stage depends upon the time-varying history of energy passing through that filter. This can be conceptualized as a buffer within which the last 30-300 ms of the waveform (passed through that filter) is stored as a vector.
  • This vector is multiplied by a second vector that is a weighting function across time.
  • Any weighting function can be used toward either biological fidelity or practical efficacy. Because the history of the signal passing through a filter is simply a vector of numbers corresponding to sampled amplitude values, the length (duration) can easily be adjusted and the function describing the weighting of energy over time is quite flexible (mathematically arbitrary). This yields a weighted history, and this weighted history of energy is converted to a single RMS value. Convolution is the preferred embodiment for applying the weighting function. The RMS value of the weighted history is then subtracted from unity (1) to yield a gain factor for that channel.
  • This simple first stage of the process mimics simple adaptation processes because the gain for the signal passing through a filter is smaller when the amount of energy that passed through the same filter in recent history is greater. In some respects, this stage is the same as that used in AGC circuits common to a many hearing aid designs; however, hearing aids with a typical small number of channels (e.g. , 4) do relatively little to change spectral properties. What distinguishes this approach is the number and design of filters. This algorithm has been implemented using 50 filters narrower than critical bands.
  • the number of filters may be increased or decreased (bandwidth decreased or increased, respectively) to optimize biological fidelity and/or practical performance criteria.
  • the duration of history also can be varied based upon fidelity or efficacy. For both bandwidth and duration, values initially may be determined on the basis of results from experiments for which frequency range and temporal parameters are investigated.
  • a second stage of processing in accordance with the invention presents another novel aspect of the approach. Because gain functions always have values ⁇ l, if all of the gain-adjusted filter outputs are simply summed (followed by biorthogonal reconstruction when filters overlap), the total energy of the output waveform will be substantially less than the input energy .
  • the simplest way to make the output level equal to the input level would be to multiply the output by a value equal to the input divided by the sum of the filter outputs.
  • the present invention provides a more sophisticated method, and permits introduction of a simulation of lateral inhibition in a fashion like that envisioned for suppression by psychoacousticians (e.g., Houtgast, T.
  • a preferred embodiment of signal processing for enhancing an auditory signal in accordance with the present invention is described with reference to Fig. 3.
  • a preferred implementation of a spectral enhancement system in a signal processor 14 or 24 passes a digitized acoustic signal x(t) to a plurality of band-pass filters 40. It is preferred that orthogonal completely-restructuring filters be used. If perfectly reconstructing filters are not used, a set of biorthogonal filters may be used to reconstruct the output signal.
  • a plurality of automatic-gain-control (AGC) circuits 41 are each associated with one band-pass filter 40.
  • the present invention targets characteristics of the acoustical signal for purposes of enhancement rather than purposes of signal compression. Time-varying gain is applied individually in each band-pass filtered signal, based on the following calculations:
  • the RMS level 42 in each individual (i ⁇ ) frequency band is calculated using a window that is determined by a function of the signal history in each frequency band.
  • This function 42 can be an arbitrary function, but the preferred embodiment is an exponential (increasing or decreasing) or a member of the exponential family.
  • Other weighting functions may be used based on practical experience with actual speech. Examples of other functions include rectangular (flat) and Gaussian (including skewed Gaussian).
  • the gain 43 in each individual (i ⁇ ) frequency band is calculated by subtracting from unity the ratio of the RMS level in the current frequency band to the sum of the RMS levels in neighboring frequency bands.
  • the neighborhood is defined by an arbitrary function. In the preferred embodiment, the neighborhood function is defined by a Gaussian function to mimic lateral interactions found in the auditory brainstem. The gain calculation results in a range of values from zero to unity, ensuring that the gain is well controlled. Additional control of the gain is accomplished by raising the gain to a specified power.
  • gain(i) [l - RMS(i)/SUM(RMS(neighborhood))]
  • the gain is applied to the signal in each individual frequency band by multiplication 44.
  • the collective effect of the windowed RMS calculation and lateral interactions within frequency neighborhoods results in a form of forward energy suppression specifically designed to enhance the spectrum of the acoustical signal.
  • This form of suppression will have the effect of sharpening dynamic modes in the spectrum, while flattening those that are relatively steady-state.
  • the output signal is obtained by summing at 45 the individually processed signals in all frequency bands. Otherwise, biorthogonal reconstructing filters will be necessary to synthesize the output signal.
  • the following is an example of the computational process for such filters which may be implemented in a computer processor using, e.g., Matlab® software.
  • f) is the jth orthogonal completely -reconstructing band-pass filter.
  • I is the length of the impulse response.
  • T is an n x n Toeplitz matrix, where n is the number of frequency channels. Neighborhoods are defined by nonzero elements of the Toeplitz matrix. Elements do not need to be unity or symmetric.

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  • Engineering & Computer Science (AREA)
  • Physics & Mathematics (AREA)
  • Health & Medical Sciences (AREA)
  • Acoustics & Sound (AREA)
  • Computational Linguistics (AREA)
  • Signal Processing (AREA)
  • Audiology, Speech & Language Pathology (AREA)
  • Human Computer Interaction (AREA)
  • Multimedia (AREA)
  • Otolaryngology (AREA)
  • Neurosurgery (AREA)
  • Computer Vision & Pattern Recognition (AREA)
  • General Health & Medical Sciences (AREA)
  • Quality & Reliability (AREA)
  • Measurement Of The Respiration, Hearing Ability, Form, And Blood Characteristics Of Living Organisms (AREA)

Abstract

Cette invention concerne une technique et un dispositif permettant d'accentuer un signal auditif dans le but de rendre des sons, en particulier des sons vocaux, plus reconnaissables. A cette fin, on subdivise un signal auditif d'entrée en une pluralité de canaux spectraux. Pour chaque canal, on obtient un gain de sortie à partir de l'évolution de l'énergie dans le temps, et de préférence, à partir de l'évolution de l'énergie dans le temps dans des canaux voisins. L'ampleur du gain de sortie ainsi obtenue pour chaque canal est de préférence inversement proportionnel à l'évolution de l'énergie dans ce même canal. Le gain de sortie obtenu pour chaque canal est appliqué au canal concerné de manière à former une pluralité de signaux de canal spectral modifié. On combine ces signaux de canal spectral modifié pour obtenir un signal auditif de sortie amélioré. La présente invention s'applique en particulier à des prothèses auditives, des systèmes de reconnaissance vocale, etc.
PCT/US2000/040854 1999-09-10 2000-09-08 Accentuation spectrale de signaux acoustiques garantissant une meilleure reconnaissance de la parole Ceased WO2001018794A1 (fr)

Priority Applications (1)

Application Number Priority Date Filing Date Title
AU12508/01A AU1250801A (en) 1999-09-10 2000-09-08 Spectral enhancement of acoustic signals to provide improved recognition of speech

Applications Claiming Priority (2)

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US15341199P 1999-09-10 1999-09-10
US60/153,411 1999-09-10

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Cited By (4)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
KR100906676B1 (ko) 2002-03-13 2009-07-08 주식회사 엘지이아이 지능형 로봇의 음성인식장치 및 방법
AU2005202837B2 (en) * 2004-06-28 2011-05-26 Hearworks Pty Limited Selective resolution speech processing
CN107004427A (zh) * 2014-12-12 2017-08-01 华为技术有限公司 增强多声道音频信号内语音分量的信号处理装置
CN110690903A (zh) * 2019-09-18 2020-01-14 南京中感微电子有限公司 一种电子设备及音频模数转换方法

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EP0556992A1 (fr) * 1992-02-14 1993-08-25 Nokia Mobile Phones Ltd. Système d'atténuation de bruit
US5388185A (en) * 1991-09-30 1995-02-07 U S West Advanced Technologies, Inc. System for adaptive processing of telephone voice signals
US5479560A (en) * 1992-10-30 1995-12-26 Technology Research Association Of Medical And Welfare Apparatus Formant detecting device and speech processing apparatus

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US5388185A (en) * 1991-09-30 1995-02-07 U S West Advanced Technologies, Inc. System for adaptive processing of telephone voice signals
EP0556992A1 (fr) * 1992-02-14 1993-08-25 Nokia Mobile Phones Ltd. Système d'atténuation de bruit
US5479560A (en) * 1992-10-30 1995-12-26 Technology Research Association Of Medical And Welfare Apparatus Formant detecting device and speech processing apparatus

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Title
SUMMERFIELD Q ET AL: "Auditory enhancement of changes in spectral amplitude", JOURNAL OF THE ACOUSTICAL SOCIETY OF AMERICA, MARCH 1987, USA, vol. 81, no. 3, pages 700 - 708, XP000978653, ISSN: 0001-4966 *

Cited By (5)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
KR100906676B1 (ko) 2002-03-13 2009-07-08 주식회사 엘지이아이 지능형 로봇의 음성인식장치 및 방법
AU2005202837B2 (en) * 2004-06-28 2011-05-26 Hearworks Pty Limited Selective resolution speech processing
CN107004427A (zh) * 2014-12-12 2017-08-01 华为技术有限公司 增强多声道音频信号内语音分量的信号处理装置
CN107004427B (zh) * 2014-12-12 2020-04-14 华为技术有限公司 增强多声道音频信号内语音分量的信号处理装置
CN110690903A (zh) * 2019-09-18 2020-01-14 南京中感微电子有限公司 一种电子设备及音频模数转换方法

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