EP2701143A1 - Sélection de modèle de conditions acoustiques pour contrôle actif du bruit - Google Patents

Sélection de modèle de conditions acoustiques pour contrôle actif du bruit Download PDF

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Publication number
EP2701143A1
EP2701143A1 EP12306013.9A EP12306013A EP2701143A1 EP 2701143 A1 EP2701143 A1 EP 2701143A1 EP 12306013 A EP12306013 A EP 12306013A EP 2701143 A1 EP2701143 A1 EP 2701143A1
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EP
European Patent Office
Prior art keywords
signal
loudspeaker
physical measurement
active noise
noise control
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EP12306013.9A
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German (de)
English (en)
Inventor
Stéphan Tassart
Fabien Ober
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OCT Circuit Technologies International Ltd
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ST Ericsson SA
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Priority to EP12306013.9A priority Critical patent/EP2701143A1/fr
Publication of EP2701143A1 publication Critical patent/EP2701143A1/fr
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    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04RLOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R3/00Circuits for transducers, loudspeakers or microphones
    • H04R3/005Circuits for transducers, loudspeakers or microphones for combining the signals of two or more microphones
    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10KSOUND-PRODUCING DEVICES; METHODS OR DEVICES FOR PROTECTING AGAINST, OR FOR DAMPING, NOISE OR OTHER ACOUSTIC WAVES IN GENERAL; ACOUSTICS NOT OTHERWISE PROVIDED FOR
    • G10K11/00Methods or devices for transmitting, conducting or directing sound in general; Methods or devices for protecting against, or for damping, noise or other acoustic waves in general
    • G10K11/16Methods or devices for protecting against, or for damping, noise or other acoustic waves in general
    • G10K11/175Methods or devices for protecting against, or for damping, noise or other acoustic waves in general using interference effects; Masking sound
    • G10K11/178Methods or devices for protecting against, or for damping, noise or other acoustic waves in general using interference effects; Masking sound by electro-acoustically regenerating the original acoustic waves in anti-phase
    • G10K11/1781Methods or devices for protecting against, or for damping, noise or other acoustic waves in general using interference effects; Masking sound by electro-acoustically regenerating the original acoustic waves in anti-phase characterised by the analysis of input or output signals, e.g. frequency range, modes, transfer functions
    • G10K11/17813Methods or devices for protecting against, or for damping, noise or other acoustic waves in general using interference effects; Masking sound by electro-acoustically regenerating the original acoustic waves in anti-phase characterised by the analysis of input or output signals, e.g. frequency range, modes, transfer functions characterised by the analysis of the acoustic paths, e.g. estimating, calibrating or testing of transfer functions or cross-terms
    • G10K11/17815Methods or devices for protecting against, or for damping, noise or other acoustic waves in general using interference effects; Masking sound by electro-acoustically regenerating the original acoustic waves in anti-phase characterised by the analysis of input or output signals, e.g. frequency range, modes, transfer functions characterised by the analysis of the acoustic paths, e.g. estimating, calibrating or testing of transfer functions or cross-terms between the reference signals and the error signals, i.e. primary path
    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10KSOUND-PRODUCING DEVICES; METHODS OR DEVICES FOR PROTECTING AGAINST, OR FOR DAMPING, NOISE OR OTHER ACOUSTIC WAVES IN GENERAL; ACOUSTICS NOT OTHERWISE PROVIDED FOR
    • G10K11/00Methods or devices for transmitting, conducting or directing sound in general; Methods or devices for protecting against, or for damping, noise or other acoustic waves in general
    • G10K11/16Methods or devices for protecting against, or for damping, noise or other acoustic waves in general
    • G10K11/175Methods or devices for protecting against, or for damping, noise or other acoustic waves in general using interference effects; Masking sound
    • G10K11/178Methods or devices for protecting against, or for damping, noise or other acoustic waves in general using interference effects; Masking sound by electro-acoustically regenerating the original acoustic waves in anti-phase
    • G10K11/1785Methods, e.g. algorithms; Devices
    • G10K11/17853Methods, e.g. algorithms; Devices of the filter
    • G10K11/17854Methods, e.g. algorithms; Devices of the filter the filter being an adaptive filter
    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10KSOUND-PRODUCING DEVICES; METHODS OR DEVICES FOR PROTECTING AGAINST, OR FOR DAMPING, NOISE OR OTHER ACOUSTIC WAVES IN GENERAL; ACOUSTICS NOT OTHERWISE PROVIDED FOR
    • G10K11/00Methods or devices for transmitting, conducting or directing sound in general; Methods or devices for protecting against, or for damping, noise or other acoustic waves in general
    • G10K11/16Methods or devices for protecting against, or for damping, noise or other acoustic waves in general
    • G10K11/175Methods or devices for protecting against, or for damping, noise or other acoustic waves in general using interference effects; Masking sound
    • G10K11/178Methods or devices for protecting against, or for damping, noise or other acoustic waves in general using interference effects; Masking sound by electro-acoustically regenerating the original acoustic waves in anti-phase
    • G10K11/1785Methods, e.g. algorithms; Devices
    • G10K11/17855Methods, e.g. algorithms; Devices for improving speed or power requirements
    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10KSOUND-PRODUCING DEVICES; METHODS OR DEVICES FOR PROTECTING AGAINST, OR FOR DAMPING, NOISE OR OTHER ACOUSTIC WAVES IN GENERAL; ACOUSTICS NOT OTHERWISE PROVIDED FOR
    • G10K11/00Methods or devices for transmitting, conducting or directing sound in general; Methods or devices for protecting against, or for damping, noise or other acoustic waves in general
    • G10K11/16Methods or devices for protecting against, or for damping, noise or other acoustic waves in general
    • G10K11/175Methods or devices for protecting against, or for damping, noise or other acoustic waves in general using interference effects; Masking sound
    • G10K11/178Methods or devices for protecting against, or for damping, noise or other acoustic waves in general using interference effects; Masking sound by electro-acoustically regenerating the original acoustic waves in anti-phase
    • G10K11/1785Methods, e.g. algorithms; Devices
    • G10K11/17857Geometric disposition, e.g. placement of microphones
    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10KSOUND-PRODUCING DEVICES; METHODS OR DEVICES FOR PROTECTING AGAINST, OR FOR DAMPING, NOISE OR OTHER ACOUSTIC WAVES IN GENERAL; ACOUSTICS NOT OTHERWISE PROVIDED FOR
    • G10K11/00Methods or devices for transmitting, conducting or directing sound in general; Methods or devices for protecting against, or for damping, noise or other acoustic waves in general
    • G10K11/16Methods or devices for protecting against, or for damping, noise or other acoustic waves in general
    • G10K11/175Methods or devices for protecting against, or for damping, noise or other acoustic waves in general using interference effects; Masking sound
    • G10K11/178Methods or devices for protecting against, or for damping, noise or other acoustic waves in general using interference effects; Masking sound by electro-acoustically regenerating the original acoustic waves in anti-phase
    • G10K11/1787General system configurations
    • G10K11/17879General system configurations using both a reference signal and an error signal
    • G10K11/17881General system configurations using both a reference signal and an error signal the reference signal being an acoustic signal, e.g. recorded with a microphone
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04RLOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R2410/00Microphones
    • H04R2410/05Noise reduction with a separate noise microphone
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04RLOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R2460/00Details of hearing devices, i.e. of ear- or headphones covered by H04R1/10 or H04R5/033 but not provided for in any of their subgroups, or of hearing aids covered by H04R25/00 but not provided for in any of its subgroups
    • H04R2460/01Hearing devices using active noise cancellation
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04RLOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R2499/00Aspects covered by H04R or H04S not otherwise provided for in their subgroups
    • H04R2499/10General applications
    • H04R2499/11Transducers incorporated or for use in hand-held devices, e.g. mobile phones, PDA's, camera's

Definitions

  • the invention relates to the field of active noise control (ANC), and more especially to the application of ANC to mobile communication terminals.
  • ANC active noise control
  • Active Noise Control also known as Active Noise Cancellation or Active Noise Reduction (ANR)
  • ANR Active Noise Reduction
  • the figure 1 illustrates these two approaches.
  • a primary source PS generates an unwanted sound.
  • This sound can be noise, or any other signal.
  • This source has been represented as a sole loudspeaker only for the clarity of the figure. In real situations, the unwanted sound can be of various natures (motors, street rum, etc.) and comes from multiple sources.
  • a loudspeaker LS has been introduced to generate the anti-noise signal. Its aim is to cancel or at least dramatically reduce the unwanted signal in a cancellation zone CZ.
  • An error microphone M E is placed within the cancellation zone CZ, in the vicinity of the loudspeaker LS in order to control the signal resulting of the emission of the anti-noise signal and the unwanted signal. It aims in capturing what is really heard by the human ear within the cancellation zone. In the ideal case, the error microphone M E measures a null signal.
  • the path between the loudspeaker LS and the error microphone M E is called the secondary path SP.
  • the simplest implementation of the feedback approach consists for the Active Noise (AN) Controller ANCS in capturing the measured signal e(n) and in re-injecting in the loudspeaker LS a signal y(n) which is a copy of e(n) with an inversion of the phase and a adaption of the amplitude.
  • AN Active Noise
  • the feed-forward approach is based on a reference microphone M R which capture a signal far enough from the loudspeaker LS so as the captured signal is not perturbed by the introduced signal.
  • the signal x(n) captured by the reference microphone M R is representative of the unwanted signal.
  • the feed-forward approach is further based on an a-priori knowledge of the transfer function of the channel linking the primary source and the cancellation zone CZ.
  • This path is equivalent to the path between the reference microphone M R and the error microphone M E , which is usually referred to as "primary path" PP.
  • the AN controller can use a filter modeling this transfer function so as to determine the re-injected signal y(n) as a function of the signal x(n) measured by the reference microphone M R .
  • the active noise controllers make use of both approach and output a re-injected signal y(n) which is function of both the signal x(n) measured by the reference microphone M R and the signal e(n) measured by the error microphone M E .
  • the AN Controller ANCS can implement various algorithm to take into account both sources of information for generating the most optimized corrective signal y(n).
  • the AN Controller ANCS should also take into account the transfer function of the secondary path SP. Whatever the chosen approach (feed-forward, feedback, or mixed), the secondary path SP should be modelized and the re-injected signal y(n) should depend on this model.
  • This secondary path SP includes the acoustic area between the loudspeaker LS and the error microphone M E which should be as close as possible of the user's ear. It also includes the transfer functions from the transducers and the audio convertors. Therefore, its model depends on the physical arrangement in which the Active Noise Control is used.
  • the physical arrangement is predetermined and fixed.
  • the model can be preset and the above-described techniques works finely. It will be the same in other situations where the secondary path linking the loudspeaker and the error microphone does not change substantially over time.
  • the user can hold the mobile terminal in various positions and press it more or less tightly on his/her ear. Also, he or she can also put it on a table, or move it from one ear to the other, etc. These variations in the acoustic characteristics of the communication channel between the loudspeaker and user's ear makes it impossible to rely on a fixed model of the secondary path SP.
  • a solution to this issue consists in estimating the model of secondary path SP in real-time, so that the AN Controller generates the corrective signal y(n) according to parameters adapted to current acoustic conditions of the secondary path.
  • the physical measurement is distinct of the at least one audio signal.
  • the at least a part of said area can be a secondary path, defined as the space between the loudspeaker and an error microphone situated inside the local zone.
  • the at least one physical measurement can comprise an electrical impedance measured on this error microphone.
  • the anti-noise signal can be generated by a loudspeaker.
  • the at least one physical measurement can comprise an indication of proximity of an object of the loudspeaker.
  • the at least a part of said area can be a primary path, defined as the space between an error microphone (M E ) situated inside said local zone (CZ), and a reference microphone situated outside the local zone, e.g. far outside the local zone.
  • M E error microphone
  • CZ local zone
  • the at least one physical measurement can comprise an estimation of the primary path.
  • Another object of the invention is an active noise control system for reducing the amount of noise in a local zone, comprising:
  • the physical measurement can be distinct of said at least one audio signal.
  • the at least a part of said area can be a secondary path, defined as the space between said loudspeaker and an error microphone situated inside said local zone.
  • the at least one physical measurement can comprise an electrical impedance measured on the error microphone.
  • the anti-noise signal can be generated by a loudspeaker.
  • the at least one physical measurement can comprise an indication of proximity of an object of the loudspeaker.
  • Another object of the invention is a computer program product comprising a computer readable medium, having thereon a computer program comprising program instructions, the computer program being loadable into a data-processing unit and adapted to cause execution of the method previously defined when the computer program is run by the data-processing unit.
  • An active noise control system ANCS is depicted on the figure 2 . It is located in front of an area A which comprises a cancellation zone CZ.
  • the cancellation zone CZ (or local zone) is the portion of space in which the active noise control is wished to produce effects.
  • the border of the cancellation zone CZ can be defined by a threshold: the effect of the cancellation/reduction of the zone inside the cancellation zone CZ is assumed to be above the threshold.
  • the cancellation zone CZ corresponds to the usage of the Active Noise Control. For instance, if the device in which the ANC is embedded is a pair of headphone, then the cancellation zone can correspond to the closed space between the ear and the headphone. If the device is a mobile communication terminal (mobile phone), then the cancellation zone is a more fuzzy space in which the user is expected to put his/her ear and the device itself.
  • the device In addition to the active noise control system ANCS, the device usually comprises several other elements, e.g. a loudspeaker SLS adapted to emit sounds related to a communication channel.
  • this loudspeaker emits the voice of the other party(ies) of the communication session. It can also emit music, vocal messages or any other audio media.
  • this loudspeaker SLS is depicted as distinct of the loudspeaker LS comprised in the active noise control system and devoted to the generation of anti-noise signal.
  • This arrangement is mainly done for the purpose of the explanations by clearly distinguishing the functions of each element.
  • both loudspeakers are implemented as a single one; this loudspeaker will emit a signal mixing the audio media and the anti-noise signal. The mixing can be done in the digital or analog electric worlds.
  • FIG. 6 it also comprises a signal microphone Ms adapted to capture the voice of the user (or any other audio signal), especially for transmission over a communication network toward to other party(ies) of a communication session.
  • a signal microphone Ms adapted to capture the voice of the user (or any other audio signal), especially for transmission over a communication network toward to other party(ies) of a communication session.
  • the mobile communication terminal T further comprises a screen D. According to the state of the art, this screen enables the user to see information provided by the terminal, but also inputs information and commands, by tactile features.
  • the reference microphone M R aims at capturing the ambient noise. Therefore, it should be placed far enough of the loudspeakers SLS and LS. It may be chosen to put it also at distance of the user's mouth.
  • the noise d which amount is to be reduced is shown as emitted by a source PS.
  • the source of this noise may not be unique. It can even be uncountable, as in the case of a rumors made of hundreds of noises which may even be of different natures. For example, in the streets, a mobile phone can be surrounded by noises coming from traffic, people's voices, music coming out from shops, etc.
  • the reference microphone M R is located close to the virtual source PS. Preferably, it should be far enough of the loudspeaker LS to capture an audio signal x that is assumed to be a good estimate for the noise d (i.e. which is not significantly perturbed by the anti-noise signal y or any other signal emitted by a signal loudspeaker SLS, not represented).
  • the error microphone M E is located close to the loudspeaker LS. It aims in capturing an error signal e, inside the cancellation area CZ, which results from the superposition of the noise signal d and the anti-noise signal generated by the loudspeaker LS. It thus provides an objective measurement of the performances of the active noise control system ANCS (i.e. how well the anti-noise signal y fully compensates for the noise signal d).
  • the error microphone M E and the reference microphone M R both capture an audio signal, respectively e, x, inside the area A. According to the invention, only one of them can be present, so as to capture only one for the audio signals e, x. According to variants of the invention, both microphones can be present, so as to capture and take benefit of these two audio signals e, x.
  • the captured audio signals e, x are transformed by the microphones in electric digital signals, respectively e(n), x(n).
  • the anti-noise signal y generated by the loudspeaker LS is function of these one or two (or, potentially more) audio signals e, y, and also of a model of the acoustic characteristics of a part of this area A.
  • the part can be the totality of this area A.
  • this part of the area A is a secondary path SP, which includes the acoustic path between the loudspeaker LS and the error microphone M E situated inside the cancellation zone CZ.
  • this part is a primary path PP defined as the space between the error microphone M E and the reference microphone M R .
  • the active noise control system ACNS comprises a sensor S for capturing at least one physical measurement s(n) representative of the acoustic characteristics of this part of the area A.
  • This physical measurement s(n), as well as the electric signals e(n), x(n) corresponding to the error and reference audio signals e, x, are transmitted to a classifier C which configures the active noise control system ANCS.
  • the anti-noise electric signal y(n) is transmitted to the loudspeaker LS which reacts by generating the corresponding audio signal y.
  • the classifier selects a model of the acoustic condition of the considered part(s) of the area A, among a set B of predetermined models. This selection is done according to at least the physical measurement s(n).
  • the anti-noise signal y(n) can dynamically follow any variations of the input signals e(n), x(n), s(n).
  • the anti-noise signal y can adapt in real-time to the changes in the acoustic conditions of the considered part(s) of the area A, in a transparent way for the user: no audio artifacts are heard even in cases of fast and important changes like when the user moves the mobile phone from the ear to put it on a table, etc.
  • the classifier C can be implemented in various ways without departing from the scope of the invention.
  • One possible embodiment consists in conforming to front-end/back-end split architecture.
  • the front end operates on the physical measurement s(n), extracts features it and group the features into a vector sent to the back end.
  • a threshold criterion applied at 1800 Hz on an estimated primary path transfer function discriminates two classes of acoustic conditions:
  • the classifier C can output two different signals y(n) to command the loudspeaker LS. This way, the presence or absence of the ear in the cancellation zone CZ is detected in real-time and can dynamically command different anti-noise signal y to adapt the situation.
  • the threshold depends on the physical parameters of the microphone, the shape of the phone etc. and should be setup by a calibration of the Active Noise Control System, or by statistical analysis of a database of measurements, for examples.
  • SVM Support Vector Machine
  • a supervised learning procedure can associate extracted features with acoustic conditions and estimated secondary path. Learning can be performed offline.
  • the SVM classifier is fast and efficient: its decisions rely mainly on the sign of different scalar products in the feature space.
  • the output of the classifier is not restricted to a discrete enumeration of recognized acoustic conditions: weighting or confidence estimates obtained from the recognition engine can be used as well in order to enable interpolation between acoustic conditions.
  • the physical measurements s(n) can be of various types also.
  • the physical measurement is distinct of the audio signals.
  • the physical measurement can be a proximity measurement provided by a proximity sensor S.
  • a proximity sensor S As proximity sensors are commonly implemented in recent mobile phones, such a solution will not add any manufacturing costs.
  • the physical measurement can be a pressure measurement provided by a mechanical pressure sensor.
  • This sensor can give an indication of the presence of the ear behind the loudspeaker LS and also discriminates between classes where the mobile phone is hold on the ear and where it is hold far from the ear.
  • the physical measurement can be a current intensity signals coming from loudspeaker LS. Indeed, it has been observed that the current intensity flowing through a loudspeaker (and its impedance) depends on the acoustic load installed in front of it. Laboratory measurements' demonstrate a strong relationship between the impedance (or current intensity), the type of obstacle (an artificial ear, in the laboratory experiments) and its relative location in front of the loudspeaker.
  • the location and the shape of the resonances are correlated with the acoustic conditions. There is no doubt that this correlation can be used in order to recognize different acoustic conditions and secondary path transfer functions.
  • the figure 3 illustrates an embodiment based on a feedback ANC and current intensity measurements.
  • the error microphone 301 captures an audio signal in the acoustic space and transform it into an electric signal through a transfer function M(s). This signal can modulate an electric carrier by a modulator 302 and is digitalized by an ADC (Analog-to-Digital Convertor) 303.
  • ADC Analog-to-Digital Convertor
  • This signal goes then through a filter 304 associated with a filter function w(z), provided by a filter selection function 311.
  • This filter minimizes the sensibility of the closed system (i.e. the rejection gain of the external noise) given the model of the acoustic conditions in front of the error microphone 301.
  • This microphone being close to the loudspeaker, these acoustic conditions are identical or substantially similar to the ones in front of the error microphone.
  • the filtered signal is then transmitted to a DAC 305 (Digital-to-Analog Convertor).
  • DAC 305 Digital-to-Analog Convertor
  • a current sensing function 306 which extracts from the signal the current intensity, without modifying substantially the signal. Such an extraction can be done by a resistance.
  • the signal sent to the loudspeaker 307 which transform the electric signal to an acoustic signal by a transfer function L(s).
  • the acoustic signal then flows through the secondary path 308, to which a transfer function S(s) can be associated.
  • This transfer function S(s) represents attenuations, echoes and other acoustic phenomena associated to geometrical features of the secondary path, obstacles, etc.
  • the resulting audio signal is superimposed with other sounds (e.g. noise) in a virtual function 309.
  • the filter w(z) still minimizes the sensibility function of the closed loop system given by the estimated model of the open loop function.
  • the filter w(z) is selected by taking as inputs the estimated impedance, which has been estimated given the signal digitalized by the ADC.
  • this current intensity is correlated by the acoustic conditions in the vicinity of the loudspeaker 307.
  • an appropriate filter can be selected among a database of available filters 312. Classification techniques mentioned earlier can be used for determining the most appropriate filter to the measured acoustic conditions.
  • the figure 4 illustrates an embodiment based on a feedback ANC and a filter selection based on a primary path estimation.
  • the block diagram of the figure 4 is similar to the block diagram of the figure 3 , with the differences of the blocks related to the capture of information used for selecting the appropriate filter.
  • the error microphone 401 captures an audio signal in the acoustic space and transform it into an electric signal through a transfer function M(s). This signal can modulate an electric carrier by a modulator 402 and is digitalized by an ADC (Analog-to-Digital Convertor) 403.
  • ADC Analog-to-Digital Convertor
  • This signal goes then through a filter 404 associated with a filter function w(z), provided by a filter selection function 411.
  • This filter minimizes the sensibility of the closed system (i.e. the rejection gain of the external noise) given the model of the acoustic conditions in front of the loudspeaker 407.
  • the filtered signal is then transmitted to a DAC 405 (Digital-to-Analog Convertor), and then to the loudspeaker 307 which transform the electric signal to an acoustic signal by a transfer function L(s).
  • DAC 405 Digital-to-Analog Convertor
  • the acoustic signal then flows through the secondary path 408, to which a transfer function S(s) can be associated.
  • This transfer function S(s) represents attenuations, echoes and other acoustic phenomena associated to geometrical features of the secondary path, obstacles, etc.
  • the resulting audio signal is superimposed with other sounds (e.g. noise) in a virtual function 409.
  • the filter w(z) still minimizes the sensibility function of the closed loop system given by the estimated model of the primary path (obtained from the adaptive feed-forward version of the ANC algorithm) or the estimation of the path between the output node of the ANC 410 and the input node of the DAC 405.
  • the reference microphone 413 has a transfer function M(s). This transfer function may be similar or different of the one of the error microphone 401.
  • the estimation of the primary path which is representative of the acoustic conditions of the primary path is then used as a criterion to select the most appropriate filter among a database of available filters 412.
  • Classification techniques mentioned earlier can be used for determining the most appropriate filter to the measured acoustic conditions.
  • the features than can be extracted from the estimated acoustic conditions of the primary path i.e. the space between the error microphone and the reference microphone. These acoustic conditions are directly related to the acoustic conditions of the pinna and to the actual secondary path transfer function; Therefore, the transfer function w(z) can be selected on this basis.
  • the figure 5 illustrates an embodiment based on an adaptive feed-forward model and filter selection based on an estimation of the current intensity.
  • the error microphone 501 captures an audio signal in the acoustic space and transform it into an electric signal through a transfer function M(s). This signal can modulate an electric carrier by a modulator 502 and is digitalized by an ADC (Analog-to-Digital Convertor) 503.
  • ADC Analog-to-Digital Convertor
  • This digital signal is sent transmitted to a LMS block 504 (for Least Mean Square).
  • the output of the LMS block corresponds to the coefficients of the FIR filter 505 associated with a time-varying transfer function w n (z), which is fed by a digital signal coming from the reference microphone 512.
  • the adaptive active noise control system ANCS is designed so as to converge as fast as possible, thanks to the LMS block 504 set before it. According to the literature, the filter w n (z) converges even in case of large variations, provided that the phase error on the secondary path is less than 90°.
  • This signal goes then from this filter 505 to a DAC 506and then to the loudspeaker 508 which transform the electric signal to an acoustic signal by a transfer function L(s).
  • an current sensing block 507 enables to extract a measurement of the current intensity without modifying substantially the signal.
  • the impedance which has been estimated given the signal digitalized by the ADC 515 and the input of the DAC 506 is used as inputs for a filter selection block 516.
  • the most appropriate filter S' (z) is selected according to this measurement among a set of available filters 517.
  • This filter S'(z) aims in estimating the transfer function S(s) associated with the secondary path. The effect of the secondary path can then be compensated.
  • This filter S'(z) has as input the digitalized measurement of a signal coming from the reference microphone 512.
  • the audio signal captured by the microphone is transformed into an electric signal by a transfer function M R (s).
  • This function can be the same of the transfer function M E (s) of the error microphone or different. It is then digitalized by the ADC 513 and then filtered by the filter 514. The filtered signal is then used as input of the LMS 504, described before.
  • the audio signal produced by the loudspeaker 508 is sur-imposed (509) with the noise flowing through the primary path 511.
  • This primary path is modelized by a transfer function P(s).

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  • Physics & Mathematics (AREA)
  • Engineering & Computer Science (AREA)
  • Acoustics & Sound (AREA)
  • Multimedia (AREA)
  • Health & Medical Sciences (AREA)
  • General Health & Medical Sciences (AREA)
  • Otolaryngology (AREA)
  • Signal Processing (AREA)
  • Soundproofing, Sound Blocking, And Sound Damping (AREA)
  • Circuit For Audible Band Transducer (AREA)
EP12306013.9A 2012-08-21 2012-08-21 Sélection de modèle de conditions acoustiques pour contrôle actif du bruit Withdrawn EP2701143A1 (fr)

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EP12306013.9A EP2701143A1 (fr) 2012-08-21 2012-08-21 Sélection de modèle de conditions acoustiques pour contrôle actif du bruit

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EP3001411A1 (fr) * 2014-09-25 2016-03-30 Eberspächer Exhaust Technology GmbH & Co. KG Protection contre les surcharges pour un acteur d'un systeme destine a influencer un bruit transporte dans une installation de gaz d'echappement
US9402132B2 (en) 2013-10-14 2016-07-26 Qualcomm Incorporated Limiting active noise cancellation output
WO2016189285A1 (fr) * 2015-05-22 2016-12-01 Cirrus Logic International Semiconductor Limited Récepteur adaptatif
CN108428444A (zh) * 2018-03-07 2018-08-21 南京大学 一种补偿次级声源近场影响的紧凑有源吸声方法
EP2755204B1 (fr) * 2013-01-15 2018-10-10 Fujitsu Limited Procédé et dispositif de suppression de bruit
WO2021204754A1 (fr) * 2020-04-07 2021-10-14 Rheinisch-Westfälische Technische Hochschule (Rwth) Aachen Procédé, dispositif, casque d'écoute et programme informatique pour supprimer activement un bruit d'interférence

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US20120057718A1 (en) * 2010-09-03 2012-03-08 Scott Dennis Vernon Noise Reduction Circuit and Method Therefor

Cited By (15)

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Publication number Priority date Publication date Assignee Title
EP2755204B1 (fr) * 2013-01-15 2018-10-10 Fujitsu Limited Procédé et dispositif de suppression de bruit
US9402132B2 (en) 2013-10-14 2016-07-26 Qualcomm Incorporated Limiting active noise cancellation output
CN105464752A (zh) * 2014-09-25 2016-04-06 埃贝赫排气技术股份有限公司 控制传播经过排气系统的声音的系统的致动器的过载保护
EP3001411A1 (fr) * 2014-09-25 2016-03-30 Eberspächer Exhaust Technology GmbH & Co. KG Protection contre les surcharges pour un acteur d'un systeme destine a influencer un bruit transporte dans une installation de gaz d'echappement
CN105464752B (zh) * 2014-09-25 2019-06-18 埃贝赫排气技术股份有限公司 控制传播经过排气系统的声音的系统的致动器的过载保护
US10215067B2 (en) 2014-09-25 2019-02-26 Eberspächer Exhaust Technology GmbH & Co. KG Overload protection for an actuator of a system for controlling sound propagating through an exhaust system
WO2016189285A1 (fr) * 2015-05-22 2016-12-01 Cirrus Logic International Semiconductor Limited Récepteur adaptatif
GB2555059A (en) * 2015-05-22 2018-04-18 Cirrus Logic Int Semiconductor Ltd Adaptive receiver
US10338883B2 (en) 2015-05-22 2019-07-02 Cirrus Logic, Inc. Adaptive receiver
US10599389B2 (en) 2015-05-22 2020-03-24 Cirrus Logic, Inc. Adaptive receiver
GB2555059B (en) * 2015-05-22 2021-09-01 Cirrus Logic Int Semiconductor Ltd Adaptive receiver
US11379176B2 (en) 2015-05-22 2022-07-05 Cirrus Logic, Inc. Adaptive receiver
CN108428444A (zh) * 2018-03-07 2018-08-21 南京大学 一种补偿次级声源近场影响的紧凑有源吸声方法
CN108428444B (zh) * 2018-03-07 2021-06-22 南京大学 一种补偿次级声源近场影响的紧凑有源吸声方法
WO2021204754A1 (fr) * 2020-04-07 2021-10-14 Rheinisch-Westfälische Technische Hochschule (Rwth) Aachen Procédé, dispositif, casque d'écoute et programme informatique pour supprimer activement un bruit d'interférence

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