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EP4356622B1 - Système de classification pour contrôle actif acoustique - Google Patents

Système de classification pour contrôle actif acoustique Download PDF

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Publication number
EP4356622B1
EP4356622B1 EP23776664.7A EP23776664A EP4356622B1 EP 4356622 B1 EP4356622 B1 EP 4356622B1 EP 23776664 A EP23776664 A EP 23776664A EP 4356622 B1 EP4356622 B1 EP 4356622B1
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Prior art keywords
filters
feedforward
filter
feedback
microphone
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EP23776664.7A
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German (de)
English (en)
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EP4356622C0 (fr
EP4356622A1 (fr
Inventor
Ludwig KOLLENZ
Michael Perkmann
Daniel Wöhrer
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Austrian Audio GmbH
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Austrian Audio GmbH
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    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R1/00—Details of transducers, loudspeakers or microphones
    • H04R1/10—Earpieces; Attachments therefor ; Earphones; Monophonic headphones
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10K—SOUND-PRODUCING DEVICES; METHODS OR DEVICES FOR PROTECTING AGAINST, OR FOR DAMPING, NOISE OR OTHER ACOUSTIC WAVES IN GENERAL; ACOUSTICS NOT OTHERWISE PROVIDED FOR
    • G10K11/00—Methods 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
    • G10K11/16—Methods or devices for protecting against, or for damping, noise or other acoustic waves in general
    • G10K11/175—Methods or devices for protecting against, or for damping, noise or other acoustic waves in general using interference effects; Masking sound
    • G10K11/178—Methods 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
    • G10K11/1781—Methods 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
    • G10K11/17813—Methods 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
    • G10K11/17815—Methods 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
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10K—SOUND-PRODUCING DEVICES; METHODS OR DEVICES FOR PROTECTING AGAINST, OR FOR DAMPING, NOISE OR OTHER ACOUSTIC WAVES IN GENERAL; ACOUSTICS NOT OTHERWISE PROVIDED FOR
    • G10K11/00—Methods 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
    • G10K11/16—Methods or devices for protecting against, or for damping, noise or other acoustic waves in general
    • G10K11/175—Methods or devices for protecting against, or for damping, noise or other acoustic waves in general using interference effects; Masking sound
    • G10K11/178—Methods 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
    • G10K11/1781—Methods 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
    • G10K11/17813—Methods 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
    • G10K11/17817—Methods 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 output signals and the error signals, i.e. secondary path
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10K—SOUND-PRODUCING DEVICES; METHODS OR DEVICES FOR PROTECTING AGAINST, OR FOR DAMPING, NOISE OR OTHER ACOUSTIC WAVES IN GENERAL; ACOUSTICS NOT OTHERWISE PROVIDED FOR
    • G10K11/00—Methods 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
    • G10K11/16—Methods or devices for protecting against, or for damping, noise or other acoustic waves in general
    • G10K11/175—Methods or devices for protecting against, or for damping, noise or other acoustic waves in general using interference effects; Masking sound
    • G10K11/178—Methods 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
    • G10K11/1783—Methods 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 handling or detecting of non-standard events or conditions, e.g. changing operating modes under specific operating conditions
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10K—SOUND-PRODUCING DEVICES; METHODS OR DEVICES FOR PROTECTING AGAINST, OR FOR DAMPING, NOISE OR OTHER ACOUSTIC WAVES IN GENERAL; ACOUSTICS NOT OTHERWISE PROVIDED FOR
    • G10K11/00—Methods 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
    • G10K11/16—Methods or devices for protecting against, or for damping, noise or other acoustic waves in general
    • G10K11/175—Methods or devices for protecting against, or for damping, noise or other acoustic waves in general using interference effects; Masking sound
    • G10K11/178—Methods 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
    • G10K11/1785—Methods, e.g. algorithms; Devices
    • G10K11/17853—Methods, e.g. algorithms; Devices of the filter
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10K—SOUND-PRODUCING DEVICES; METHODS OR DEVICES FOR PROTECTING AGAINST, OR FOR DAMPING, NOISE OR OTHER ACOUSTIC WAVES IN GENERAL; ACOUSTICS NOT OTHERWISE PROVIDED FOR
    • G10K11/00—Methods 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
    • G10K11/16—Methods or devices for protecting against, or for damping, noise or other acoustic waves in general
    • G10K11/175—Methods or devices for protecting against, or for damping, noise or other acoustic waves in general using interference effects; Masking sound
    • G10K11/178—Methods 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
    • G10K11/1787—General system configurations
    • G10K11/17879—General system configurations using both a reference signal and an error signal
    • G10K11/17881—General system configurations using both a reference signal and an error signal the reference signal being an acoustic signal, e.g. recorded with a microphone
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10K—SOUND-PRODUCING DEVICES; METHODS OR DEVICES FOR PROTECTING AGAINST, OR FOR DAMPING, NOISE OR OTHER ACOUSTIC WAVES IN GENERAL; ACOUSTICS NOT OTHERWISE PROVIDED FOR
    • G10K11/00—Methods 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
    • G10K11/16—Methods or devices for protecting against, or for damping, noise or other acoustic waves in general
    • G10K11/175—Methods or devices for protecting against, or for damping, noise or other acoustic waves in general using interference effects; Masking sound
    • G10K11/178—Methods 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
    • G10K11/1787—General system configurations
    • G10K11/17885—General system configurations additionally using a desired external signal, e.g. pass-through audio such as music or speech
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10K—SOUND-PRODUCING DEVICES; METHODS OR DEVICES FOR PROTECTING AGAINST, OR FOR DAMPING, NOISE OR OTHER ACOUSTIC WAVES IN GENERAL; ACOUSTICS NOT OTHERWISE PROVIDED FOR
    • G10K2210/00—Details of active noise control [ANC] covered by G10K11/178 but not provided for in any of its subgroups
    • G10K2210/10—Applications
    • G10K2210/108—Communication systems, e.g. where useful sound is kept and noise is cancelled
    • G10K2210/1081—Earphones, e.g. for telephones, ear protectors or headsets
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10K—SOUND-PRODUCING DEVICES; METHODS OR DEVICES FOR PROTECTING AGAINST, OR FOR DAMPING, NOISE OR OTHER ACOUSTIC WAVES IN GENERAL; ACOUSTICS NOT OTHERWISE PROVIDED FOR
    • G10K2210/00—Details of active noise control [ANC] covered by G10K11/178 but not provided for in any of its subgroups
    • G10K2210/30—Means
    • G10K2210/301—Computational
    • G10K2210/3038—Neural networks
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10K—SOUND-PRODUCING DEVICES; METHODS OR DEVICES FOR PROTECTING AGAINST, OR FOR DAMPING, NOISE OR OTHER ACOUSTIC WAVES IN GENERAL; ACOUSTICS NOT OTHERWISE PROVIDED FOR
    • G10K2210/00—Details of active noise control [ANC] covered by G10K11/178 but not provided for in any of its subgroups
    • G10K2210/30—Means
    • G10K2210/301—Computational
    • G10K2210/3041—Offline
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10K—SOUND-PRODUCING DEVICES; METHODS OR DEVICES FOR PROTECTING AGAINST, OR FOR DAMPING, NOISE OR OTHER ACOUSTIC WAVES IN GENERAL; ACOUSTICS NOT OTHERWISE PROVIDED FOR
    • G10K2210/00—Details of active noise control [ANC] covered by G10K11/178 but not provided for in any of its subgroups
    • G10K2210/30—Means
    • G10K2210/301—Computational
    • G10K2210/3048—Pretraining, e.g. to identify transfer functions
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10K—SOUND-PRODUCING DEVICES; METHODS OR DEVICES FOR PROTECTING AGAINST, OR FOR DAMPING, NOISE OR OTHER ACOUSTIC WAVES IN GENERAL; ACOUSTICS NOT OTHERWISE PROVIDED FOR
    • G10K2210/00—Details of active noise control [ANC] covered by G10K11/178 but not provided for in any of its subgroups
    • G10K2210/30—Means
    • G10K2210/301—Computational
    • G10K2210/3056—Variable gain
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R2460/00—Details 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
    • H04R2460/01—Hearing devices using active noise cancellation
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R2460/00—Details 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
    • H04R2460/15—Determination of the acoustic seal of ear moulds or ear tips of hearing devices

Definitions

  • the invention relates to a method for classifying and applying filters for active noise control in hearing systems according to claim 1.
  • Hearing systems of all kinds be they hearing aids or headphones (hereinafter used as a synonym for all types of hearing systems) of the types over-ear, on-ear, in-ear, or ear buds, with ANC (used in this application as an abbreviation for Active Noise Cancelling or Active Noise Control and synonymously with ANR, i.e. Active Noise Reduction) are affected by the problem of ANC performance depending on the wearing situation. Ears are very different from person to person and the wearing situation changes every time the headphones are put on or inserted, which has a major impact on the ANC performance, especially in the case of static, i.e. non-adaptive systems.
  • ANC used in this application as an abbreviation for Active Noise Cancelling or Active Noise Control and synonymously with ANR, i.e. Active Noise Reduction
  • headphones do not usually have the same passive attenuation everywhere and in every wearing situation, which is why the passive attenuation of the headphones varies depending on the direction of incidence, and thus also the ANC performance.
  • a system must evaluate the current wearing situation/incident direction of noise and adjust the filters of the ANC circuit.
  • a typical approach is adaptive filters, with Least Mean Squares (LMS) being the most common.
  • LMS Least Mean Squares
  • Adaptive control usually requires an identical sampling rate for the filter and the LMS algorithm. However, this is not usually the case in integrated circuits (ICs), as audio processing takes place at high sampling rates (e.g. 384 kHz) and control at low sampling rates (e.g.
  • the US9,773,490 B2 discloses a method in which an acoustic leakage between a speaker of a headphone and an error microphone is measured or estimated and the feedback system is adjusted accordingly in order to avoid instabilities in the ANC system.
  • the basis for this method is the presence of a useful signal (source signal) which is measured at the reference microphone.
  • source signal a useful signal which is measured at the reference microphone.
  • the US 2022/0076656 A1 discloses another method for estimating the leakage of an ANC headphone using feedforward and feedback microphone signals and corresponding selection of ANC profiles.
  • the US9,142,205 B2 further discloses a method that measures or estimates the acoustic leakage between the loudspeaker and the error microphone and adjusts the feedback ANC system so that part of the playback signal that arrives at the reference microphone is not canceled out.
  • the disadvantage of this method is that it only optimizes playback performance for the ideal useful signal.
  • the US9,516,407 B2 shows a method for estimating the transfer function of an ANC system.
  • One of the disadvantages is the complex two-stage process of filter selection.
  • a method for classifying and applying ANC systems in headphones which has the method steps specified in claim 1.
  • a method is used that identifies the wearing situation and direction of interference noise of ANC headphones as well as the physical properties (e.g. ear shape, jaw/skull shape,...) of the person wearing the headphones and, by means of a classifying algorithm, selects the filter best suited to the respective wearing situation and noise environment from a large number of filters, so that it can be used for the ANC system.
  • a key advantage of the method is that the ANC system can use sampling rates independently of the system selecting the filter. The independent implementation between the filter selecting system part and the ANC system part enables a more energy efficient and error resistant implementation.
  • measurements must be carried out for headphones that use the method according to the invention and the transmission distances for different ears (which are shaped differently for each wearer), wearing situations and directions of noise in different environments must be determined.
  • These measurements are carried out in a controlled environment (e.g. acoustic laboratory) on one or more headphones of the same model.
  • a set of filters is determined that covers the different situations (or averages of these). This set of filters represents the widest possible range of situations in which noise can occur, from street noise to turbine noise, the background noise of a coffee house to children playing. The technicians have hardly any limits here due to the controlled environment of the acoustic laboratory.
  • the application of the pre-filters from point a1) is optional.
  • the FF or FB paths must also be determined analogously for each of the desired combinations. This step is logical for the person skilled in the art with knowledge of the invention and can be carried out without further explanation. However, in the above description of the method, the singular number of components was chosen to make the example easier to read.
  • the gain of the filter can optionally be adjusted either before, in parallel, or after filter selection. It should be noted that the estimated and applied transfer functions can be different.
  • the filter coefficients assigned to the class are applied in the ANC path.
  • the filter based on the selected coefficients can be the right choice in different wearing situations as long as the gain is adjusted correctly. This can lead to the case that, mathematically speaking, another filter would be more suitable, but the selected filter is sufficiently strong due to its gain not to trigger an adjustment of the filter. In this case, there is therefore an interplay between the gain estimator and the classifier.
  • a gain estimator can be a PID controller, LMS (with one coefficient), or cross-correlation.
  • the coefficient set can be used in addition to the FF and FB filters already mentioned (in Fig.1 represented as FF(z) and FB(z)) also contain different coefficients for audio playback.
  • the audio playback path is not only dependent on the audio source (audio stream source in Fig.1 ), but also by the wearing situation and the physical characteristics (e.g. ear shape, jaw/skull shape,...) of the person wearing the headphones and should ideally be adjusted.
  • This adjustable filter for audio playback is in Fig.1 displayed as audio(z).
  • All of these described audio chains are brought together via a mixer before playback. This can be equipped with a variable gain just like the individual paths.
  • the final playback of the audio chain can be done via various systems such as dynamic speakers, balanced armature drivers, MEMS speakers, bone-conduction systems, etc. and is in Fig.1 represented as a loudspeaker.
  • the selection of the filter function and the calculation of the gain can be carried out simultaneously or alternately (ping-pong mode). Simultaneously is understood here to mean that the processor carrying out the calculation simultaneously makes the calculation results available to the system at the end of the calculation cycle.
  • the duration of a calculation cycle with alternating selection of the filter function and calculation of the gain can be selected variably.
  • the system can allow the estimate of the transfer function to converge for 100 ms, then adjust the gain for 100 ms, then estimate the transfer function again, etc. Variations of this scheme are easily understandable and feasible for the person skilled in the art with knowledge of the invention.
  • a gain estimator from step II can be a PID controller known from control engineering, an LMS algorithm (with a coefficient) or a cross-correlation function (see above).
  • step IV stopping or starting the adaptation
  • a noise floor for each of the different parts of the system, i.e. a lower threshold which can be given as an absolute limit by the sensor sensitivity, or which can be firmly defined for the algorithm. If this threshold is exceeded, the adaptation is stopped.
  • An example of this is a quiet environment: Since there is hardly any energy in the acoustic signal, only poor estimates can be made, and good ANC performance is not relevant because the environment is quiet anyway.
  • the adaptation of the filters is only paused if the performance (indicated by the performance indicator) is sufficiently good or if disturbances are detected.
  • the threshold values for starting and stopping the adaptation based on the indicator can be static or adaptive as already described and in the latter case change with a long-term averaging of the ANC overall performance.
  • the algorithm tries to achieve a noise minimum.
  • Stopping the adaptation is necessary to prevent borderline cases in which the classifier would constantly switch between two states (for example two filters).
  • Pausing in the event of interference is then advantageous to prevent incorrect adaptation.
  • a reference threshold is set to which the smoothed signal is fed.
  • the decision as to whether adaptation takes place is made by a threshold switch, which usually has a hysteresis.
  • the performance indicator thus also prevents jumping between filters too often.
  • the classifier can be set to pay attention to the current value of the gain estimate: If the value of the gain estimate is at its maximum or minimum over a defined period of time, the classifier can switch to the next higher or lower filter class. The reason for this is that, for example, with a constantly high gain, a neighboring filter probably delivers better results with a medium gain, which was not reflected in the transfer function estimation.
  • the classifier only pays attention to the gain estimate, since in the extreme case the transfer function can only be made dependent on the gain estimate, and switches classes according to this information.
  • Fig.3 shows the flow chart of the example Fig.1 and Fig. 2 used classifier with an optional timer, which is used for the ping-pong variants.
  • the timer has a duty cycle, similar to a periodic square function, and fulfills the task of switching between gain estimation and classifier.
  • a timer designed in an analogous manner can also be used to control the sequence of adaptation runs for FB and FF filters (point l) in the description of Fig.1 ).
  • Fig.4 shows a flow chart showing the filter selection by the classifying algorithm (classifier), whereby the scheme only applies from subsection g) of the procedure on the first run.
  • an adaptive algorithm (see point b) or h)) is used to estimate the transfer functions of the FF path and/or the FB path used and a performance indicator is used, which is determined by referencing the feedback microphone signal to the feedforward microphone signal (see point i)).
  • the transfer functions estimated by the adaptive algorithm and the performance indicator serve as the input of the classifying algorithm.
  • the estimated transfer function provides the classifying algorithm with the characteristics of the current wearing situation based on the energy content of the ambient sound, i.e. it "recognizes" the situation and tries to select the right filter based on the transfer function.
  • the performance indicator shows the classifying algorithm whether the selected filter is sufficiently suitable for the analyzed and classified situation, or whether a better filter must be selected.
  • the classifier processes both parameters together. This is done by the classifying algorithm, such as a neural network, mapping the mismatch between the measuring point (FB microphone) and the target point (eardrum) based on its prior training. This information is used to identify which suitable filter is assigned to which mismatch based on the transfer function estimated by the adaptive algorithm.
  • the determined filter is taken from the lookup table (LUT) stored on the IC.
  • the filter taken from the LUT is then applied in the FF or FB path.
  • the advantage of using such a classifier over adaptive filters is the ability to carry out measurements in advance (e.g. in special laboratories by a specialist) that allow for different conditions, such as a low-reflection room or a diffuse sound field.
  • the filters can be calculated for sampling rates other than the sampling rate of the classifier. It is common for classifiers to work with sampling rates ⁇ 50 kHz, while filters with >300 kHz are used.
  • Another major advantage of the invention over an adaptive system that calculates the filters in real time is the ability to determine the filters, for example, with in-situ measurements, i.e. with probe microphones close to the eardrum (in this case the target point).
  • the LUT is separated from the transfer function estimation and any classification can be carried out.
  • the state of the art US9,516,407 B2
  • the transfer function estimation is limited to its reference points: for example, the microphones (feedforward and feedback).
  • the filters in the LUT can be created for any target point, which does not necessarily have to correspond to the microphone points.
  • the classifier is trained to map the difference between the microphone point and the target point.
  • the filters in the LUT are designed for the target point (as close as possible to the drum field point).
  • the transfer function estimation is optimized for the feedback microphone point.
  • the classifier is trained in a laboratory environment using in-situ measurements for the target point in reference to the microphone points (feedforward and feedback).
  • the filters are arranged hierarchically according to class in a filter matrix depending on shape and gain. This allows sequential jumps and allows the algorithm to vary in borderline cases between two filters with slightly different shapes or between slightly different gains.
  • a major advantage of this method over adaptive filters is that only useful filters are stored in memory.
  • the ANC algorithm therefore does not run the risk of getting stuck at local optima.
  • the system is therefore stable in any case and does not generate any artifacts (e.g. hiss noise).
  • a particularly preferred embodiment of this method uses a combination of a classical LMS algorithm and a PEAK filter in the sense of the application instead of just a classical LMS algorithm EP2022/068392 , published as WO2023280752A1 on January 12, 2023. In this way, the advantages of the cited application are transferred to the application presented here.
  • an audio playback filter can be assigned to each class (in addition to feedforward and/or feedback filters). This is intended for playing an audio source (e.g. Bluetooth audio or 3.5mm jack) and has the task of keeping the sound image constant when the wearing situation changes.
  • an audio source e.g. Bluetooth audio or 3.5mm jack
  • the feedback path is defined as a transmission path described as a transfer function between an internal feedback microphone located near the loudspeaker and the loudspeaker.
  • the feedforward path is defined as the calculated transmission distance (transfer function) based on the transfer function of the FF microphone, the internal transfer function of the loudspeakers (headphone speaker frequency response) and the transfer function of the passive damping (mechanical system).
  • the invention relates to a method for classifying and applying acoustic filters for active noise control in hearing systems, whereby the filters are determined in advance and stored in a memory in the headphones. While wearing the headphones, it is thus possible to quickly and efficiently select and apply a specific filter in order to improve ANC performance and stability.

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  • Physics & Mathematics (AREA)
  • Engineering & Computer Science (AREA)
  • Acoustics & Sound (AREA)
  • Multimedia (AREA)
  • Signal Processing (AREA)
  • Health & Medical Sciences (AREA)
  • Audiology, Speech & Language Pathology (AREA)
  • General Health & Medical Sciences (AREA)
  • Soundproofing, Sound Blocking, And Sound Damping (AREA)

Claims (7)

  1. Procédé de classification et d'application de filtres pour le contrôle actif du bruit dans des systèmes d'écoute comprenant au moins un microphone feedforward et au moins un microphone feedback, au moins un haut-parleur, au moins un circuit intégré constitué d'une mémoire et d'au moins une unité de processeur, et comprenant au moins un chemin feedback, décrit par une fonction de transfert, et au moins un chemin feedforward décrit par une fonction de transfert, comprenant les étapes :
    a) de réception d'un signal audio désigné comme signal de microphone feedback, par l'intermédiaire du microphone feedback, au nombre d'au moins, et d'un signal audio désigné comme signal de microphone feedforward, par l'intermédiaire du microphone feedforward, au nombre d'au moins un,
    b) d'estimation de la fonction de transfert du chemin feedback par un algorithme adaptatif, sur la base de la comparaison du signal de microphone feedback et d'un signal de haut-parleur,
    c) de calcul de différentes caractéristiques pertinentes, pas nécessairement de l'ensemble des caractéristiques mais d'au moins une, à partir de la fonction de transfert du chemin feedback estimée sous b),
    d) de détermination du niveau sur le microphone feedback par rapport au microphone feedforward,
    e) où le choix d'un jeu de coefficients par un algorithme de classification, sur la base des caractéristiques pertinentes déterminées sous c) et de la relation du niveau sur le microphone feedback par rapport au microphone feedforward déterminée sous d), s'effectue soit par émission d'une classe discrète qui est associée à un jeu de filtres stocké dans la mémoire située dans le casque, soit par émission d'une fonction à une dimension qui choisit un jeu de filtres stocké dans la mémoire située dans le casque,
    f) d'application du jeu de coefficients du filtre choisi sous e), dans le chemin audio feedback actuel,
    g) d'estimation de la fonction de transfert du chemin feedforward, par comparaison du signal de microphone feedforward et feedback déterminé sous a), en utilisant un algorithme adaptatif pour l'estimation de la fonction de transfert du chemin feedforward,
    h) de calcul de différentes caractéristiques pertinentes, pas nécessairement de l'ensemble des caractéristiques mais d'au moins une, à partir de la fonction de transfert du chemin feedforward estimée sous g),
    i) de détermination d'un indicateur de performance, par référencement du signal de microphone feedback déterminé sous a) vers le signal de microphone feedforward,
    j) d'utilisation de la fonction de transfert estimée sous h) et de l'indicateur de performance déterminé sous i), en tant qu'entrées d'un algorithme de classification qui choisit ainsi un jeu de coefficients, soit par émission d'une classe discrète qui est associée à un jeu de filtres stocké dans la mémoire située dans le casque, soit par émission d'une fonction à une dimension qui choisit un jeu de filtres stocké dans la mémoire située dans le casque,
    k) d'application du jeu de coefficients du filtre choisi sous j), dans le chemin audio feedforward actuel,
    l) et de réajustement des filtres par une nouvelle exécution du procédé, soit en exécutant de nouveau la totalité du procédé à partir du point a), soit en procédant seulement à une adaptation du filtre feedforward, c'est-à-dire en exécutant le procédé à partir du point g).
  2. Procédé de classification et d'application de filtres pour le contrôle actif du bruit dans des systèmes d'écoute selon la revendication 1, caractérisé en ce que, après réception du signal audio par le microphone feedback, on réalise un préfiltrage du signal audio à l'aide d'un filtre passe-haut, passe-bas, passe-bande ou un filtre similaire.
  3. Procédé de classification et d'application de filtres pour le contrôle actif du bruit dans des systèmes d'écoute selon la revendication 1 ou 2, caractérisé en ce que, après réception du signal audio par le microphone feedforward, on réalise un préfiltrage du signal audio à l'aide d'un filtre passe-haut, passe-bas, passe-bande ou un filtre similaire.
  4. Procédé de classification et d'application de filtres pour le contrôle actif du bruit dans des systèmes d'écoute selon une des revendications 1 à 3, caractérisé en ce que, après une première adaptation du filtre feedback et feedforward, l'amplification du filtre peut être adaptée.
  5. Procédé de classification et d'application de filtres pour le contrôle actif du bruit dans des systèmes d'écoute selon une des revendications 1 à 4, caractérisé en ce que le niveau sur le microphone feedback et/ou l'indicateur de performance est utilisé pour déterminer l'amplification du filtre qui est idéale pour la situation de port.
  6. Procédé de classification et d'application de filtres pour le contrôle actif du bruit dans des systèmes d'écoute selon une des revendications 1 à 5, caractérisé en ce qu'est défini pour le système, un filtre par défaut qui est utilisé en cas de passage en dessous d'une sortie de capteur prédéfinie du système ANC.
  7. Procédé de classification et d'application de filtres pour le contrôle actif du bruit dans des systèmes d'écoute selon une des revendications 1 à 6, caractérisé en ce que les filtres font l'objet d'un tri hiérarchique par classes, en fonction de la forme et du gain, et sont disposés dans une matrice de filtres.
EP23776664.7A 2022-09-30 2023-09-29 Système de classification pour contrôle actif acoustique Active EP4356622B1 (fr)

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US9142205B2 (en) 2012-04-26 2015-09-22 Cirrus Logic, Inc. Leakage-modeling adaptive noise canceling for earspeakers
US9123321B2 (en) 2012-05-10 2015-09-01 Cirrus Logic, Inc. Sequenced adaptation of anti-noise generator response and secondary path response in an adaptive noise canceling system
US9516407B2 (en) 2012-08-13 2016-12-06 Apple Inc. Active noise control with compensation for error sensing at the eardrum
US11842717B2 (en) * 2020-09-10 2023-12-12 Maxim Integrated Products, Inc. Robust open-ear ambient sound control with leakage detection
US11303258B1 (en) * 2020-09-16 2022-04-12 Apple Inc. Method and system for adaptive audio filters for different headset cushions
US11468875B2 (en) * 2020-12-15 2022-10-11 Google Llc Ambient detector for dual mode ANC
EP4117306A1 (fr) 2021-07-05 2023-01-11 Austrian Audio GmbH Procédé électro-acoustique utilisant un algorithme lms

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