WO2025196331A1 - Procédé de traitement de signaux audio - Google Patents

Procédé de traitement de signaux audio

Info

Publication number
WO2025196331A1
WO2025196331A1 PCT/EP2025/057939 EP2025057939W WO2025196331A1 WO 2025196331 A1 WO2025196331 A1 WO 2025196331A1 EP 2025057939 W EP2025057939 W EP 2025057939W WO 2025196331 A1 WO2025196331 A1 WO 2025196331A1
Authority
WO
WIPO (PCT)
Prior art keywords
filter
audio signal
pass
function
low
Prior art date
Legal status (The legal status is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the status listed.)
Pending
Application number
PCT/EP2025/057939
Other languages
German (de)
English (en)
Inventor
Johannes Fabry
Raphael Nicolas BRANDIS
Stefan Liebich
Current Assignee (The listed assignees may be inaccurate. Google has not performed a legal analysis and makes no representation or warranty as to the accuracy of the list.)
Elevear GmbH
Original Assignee
Elevear GmbH
Priority date (The priority date is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the date listed.)
Filing date
Publication date
Application filed by Elevear GmbH filed Critical Elevear GmbH
Publication of WO2025196331A1 publication Critical patent/WO2025196331A1/fr
Pending legal-status Critical Current
Anticipated expiration legal-status Critical

Links

Classifications

    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; ELECTRIC HEARING AIDS; PUBLIC ADDRESS SYSTEMS
    • H04R3/00—Circuits for transducers
    • H04R3/04—Circuits for transducers for correcting frequency response
    • H—ELECTRICITY
    • H03—ELECTRONIC CIRCUITRY
    • H03G—CONTROL OF AMPLIFICATION
    • H03G5/00—Tone control or bandwidth control in amplifiers
    • H03G5/16—Automatic control
    • H03G5/165—Equalizers; Volume or gain control in limited frequency bands

Definitions

  • the present invention relates to a method for processing audio signals and a corresponding device.
  • Various methods for audio signal processing are known from the prior art for different application scenarios.
  • DE 10 2022111300 A1 and US Pat. No. 6,141,672 A describe methods for (audio) signal processing.
  • the methods known from the prior art are used in particular to suppress background noise in the field of communications technology and in hearing aids.
  • unfiltered audio signals are often recorded and then converted to the frequency domain before the audio signal is filtered in the frequency domain.
  • the short-time Fourier transform (STFT) is typically used to convert the recorded time signal to the frequency domain.
  • STFT short-time Fourier transform
  • the resulting latency can lead to an unnatural perception or even disorientation of a user, since the perceived acoustic signals are less pronounced than the visual perception (for example, in the case of a March 14, 2025 Conversation between two people, one of whom wears a hearing aid) is delayed.
  • the superposition of passive and delayed, actively reproduced sound can result in comb filter effects that distort the frequency spectrum. If the processing latency is also too high, the system cannot react quickly enough to impulse noises, for example, to protect hearing from damage.
  • Many of the available audio codecs have dedicated processors for recording, processing, and playing back audio signals with low latency. The conversion and processing of signals is often performed per sampling point, so that the latency can be in the low microsecond range at correspondingly high sampling rates.
  • - two high-pass filters in particular two adjacent high-pass filters
  • the high-pass filter transfer function ⁇ ( ⁇ ) of which is designed such that their magnitude response above a first cut-off frequency ⁇ ⁇ , ⁇ has a deviation of maximum 10% from one another, preferably a deviation of maximum 5% and particularly preferably a deviation of maximum 3% or maximum 1%
  • - two low-pass filters in particular two adjacent low-pass filters
  • the low-pass filter transfer function ⁇ ( ⁇ ) of which is designed such that their magnitude response below a second cut-off frequency ⁇ ⁇ , ⁇ has a deviation of maximum 10% from one another, preferably a deviation of maximum 5% and particularly preferably a deviation of maximum 3% or maximum 1%.
  • two high-pass filters can be used, designed so that their magnitude responses approach each other at high frequencies, or two low-pass filters can be used, designed so that their magnitude responses approach each other at low frequencies.
  • the first cutoff frequency and the second cutoff frequency are also referred to as stopband frequencies in the context of the present invention.
  • the deviation of the magnitude responses (in percent) for two high-pass filters or two low-pass filters can be defined as follows: where ⁇ denotes the deviation of the magnitude responses,
  • the deviation ⁇ ⁇ of the magnitude responses above or below the corresponding stopband frequency is, in the preferred embodiments , always less than the aforementioned 10%, 5%, 3%, or 1 %.
  • - two high-pass filters are used, the high-pass filter transfer function ⁇ ( ⁇ ) of which is each designed such that their phase response above a first cut-off frequency ⁇ , ⁇ has a deviation of a maximum of 10% from each other March 14, 2025, preferably a deviation of a maximum of 5% and particularly preferably a deviation of a maximum of 3% or a maximum of 1%, or - two low-pass filters are used whose low-pass filter transfer function ⁇ ( ⁇ ) is designed such that their phase response below a second cutoff frequency ⁇ ⁇ , ⁇ has a deviation of a maximum of 10% from each other, preferably a deviation of a maximum of 5% and particularly preferably a deviation of 3% or a maximum of 1%.
  • two high-pass filters can be used that are designed such that their phase response approximates at high frequencies
  • two low-pass filters can be used that are designed such that their phase response approximates at low frequencies.
  • the deviation of the phase responses (in percent) for two high-pass filters or two low-pass filters can be defined analogously to the deviation of the magnitude responses described above: 100, where ⁇ is the deviation of the phase responses, the phase response of a first individual filter (high-pass filter or low-pass filter), and ⁇ ⁇ the phase response of a second individual filter (high-pass filter or low-pass filter) above or below the corresponding stopband frequency.
  • the deviation ⁇ ⁇ of the phase responses above or below the corresponding stopband frequency is, in the preferred embodiments, always less than the aforementioned 10%, 5%, 3% or 1%.
  • FIR filters are always stable because they do not use feedback loops.
  • the use of IIR filters has the advantage that they are highly efficient .
  • sharper cutoff frequencies can be achieved by using IIR filters compared to FIR filters.
  • the method according to the invention can provide for the individual filters to have a non-linear phase response. Initial studies have shown that a further reduction in latency can be achieved by using individual filters with a non-linear phase response.
  • the filter function allows efficient processing of the audio signals using high-pass filters, thus enabling low latency.
  • the filtered audio signal ⁇ ( ⁇ ) is determined as follows : where ⁇ ( ⁇ ) denotes a weighting factor and ⁇ ( ⁇ ) is determined as follows : where ⁇ ( ⁇ ) each describes an audio signal ⁇ ( ⁇ ) filtered by a low-pass filter ⁇ , and where the low-pass filters ⁇ each have a cutoff frequency ⁇ , ⁇ at which ⁇ , ⁇ ⁇ ⁇ , ⁇ .
  • the implementation of the filter function described above enables efficient processing of the audio signals using low-pass filters, thereby enabling low latency.
  • the subtraction of the individual filter functions can also be achieved implicitly using modified weighting factors.
  • the filtered audio signal ⁇ ( ⁇ ) is determined as follows: where ⁇ ( ⁇ ) describes an audio signal ⁇ ( ⁇ ) filtered by a high-pass filter ⁇ , where the high-pass filters ⁇ each have a cutoff frequency March 14, 2025 ⁇ , ⁇ where ⁇ , ⁇ ⁇ ⁇ , ⁇ , and ⁇ ( ⁇ ) each denotes a modified gain factor determined as follows: . This allows for a particularly efficient implementation of the filter function using high-pass filters, thereby achieving low latency.
  • the filtered audio signal ⁇ ( ⁇ ) is determined as follows: where ⁇ ( ⁇ ) each describes an audio signal ⁇ ( ⁇ ) filtered by a low-pass filter ⁇ , and where the low-pass filters ⁇ each have a cutoff frequency ⁇ , ⁇ at which ⁇ , ⁇ ⁇ ⁇ , ⁇ applies, and ⁇ ( ⁇ ) each denotes a modified gain factor , which is determined as follows: This allows a particularly efficient implementation of the filter function using low-pass filters to be provided, thereby enabling low latency.
  • at least one individual filter function is designed as a delta function (also referred to as a Dirac function or unit impulse function).
  • the second frequency ( ⁇ ⁇ ) or frequency with which the weighting factors or the modified weighting factors are calculated can be chosen to be lower than the first frequency ( ⁇ ⁇ ) or the frequency with which the multiplication of the ( modified) weighting factors with the filtered audio signals ⁇ ( ⁇ ) is carried out, as well as the summation of the products of the (modified) weighting factors with the filtered audio signals ⁇ ⁇ ( ⁇ ) , without the quality of the filtered audio signals being significantly impaired.
  • This allows for more efficient processing of the audio signals without the March 14, 2025 quality of the filtered audio signals. As a result, a further reduction in latency can be achieved.
  • the first frequency 192 ⁇
  • the first frequency and the second frequency can be modified.
  • the ratio ⁇ ⁇ between and ⁇ ⁇ are integers.
  • the ratio can be selected according to requirements, advantageously with a value between 2 and 64, preferably with a value between 4 and 32, particularly preferably with a value between 8 and 16.
  • the second frequency ⁇ ⁇ can preferably be ⁇ 48 kHz, ⁇ 24 kHz or ⁇ 16 kHz.
  • a device for processing audio signals comprising: - a recording unit for recording an audio signal ⁇ ( ⁇ ); - a computing unit for calculating a filter function and for processing the audio signal ⁇ ( ⁇ ); wherein the computing unit is designed to apply the filter function to the audio signal ⁇ ( ⁇ ) in the time domain and to calculate a filtered audio signal ⁇ ( ⁇ ); - an output unit for outputting the filtered audio signal ⁇ ( ⁇ ); characterized in that March 14, 2025 - the computing unit is designed to determine the filter function as a function of ⁇ individual filter functions, where ⁇ ⁇ 2; and - to determine a bandpass filter function from at least two individual filter functions, which comprise two high-pass filter functions or two low-pass filter functions.
  • the device according to the invention allows particularly efficient processing of audio signals with low latency.
  • the device according to the invention can be provided with the computing unit being designed to calculate a bandpass filter function by subtracting a first individual filter function from a second individual filter function, wherein the first individual filter function and the second individual filter function are each embodied as a filter function of a high-pass filter or a filter function of a low-pass filter.
  • the device according to the invention can be provided with the properties described above in connection with the method according to the invention and with the computing unit of the device according to the invention being designed to carry out the method steps described in connection with the method.
  • the high-pass filters or the low-pass filters used are designed as non-linear-phase filters.
  • the present invention can provide for the high-pass filters or the low-pass filters to be designed as recursive, minimum-phase filters.
  • March 14, 2025 it can preferably be provided that the weighting factors or the modified weighting factors are calculated as a function of at least one sensor signal, in particular as a function of a microphone signal.
  • FIG. 1 shows a flow diagram for an embodiment of the present invention
  • FIG. 2 shows a signal flow diagram for an embodiment of the present invention
  • FIG. 3 shows an exemplary magnitude response of high-pass filters according to an embodiment of the present invention
  • FIG. 4 shows an exemplary phase response of high-pass filters according to an embodiment of the present invention
  • FIG. 5 shows an exemplary magnitude response of resulting band-pass filters according to an embodiment of the present invention
  • FIG. 6 shows a signal flow diagram for an embodiment of the present invention based on low-pass filters
  • FIG. 1 shows a flow diagram for an embodiment of the present invention
  • FIG. 2 shows a signal flow diagram for an embodiment of the present invention
  • FIG. 3 shows an exemplary magnitude response of high-pass filters according to an embodiment of the present invention
  • FIG. 4 shows an exemplary phase response of high-pass filters according to an embodiment of the present invention
  • FIG. 5 shows an exemplary magnitude response of resulting band-pass filters according to an embodiment of the present invention
  • FIG. 6 shows a signal flow diagram for an
  • a filter function is provided.
  • the filter function is provided as a function of ⁇ individual filter functions, where ⁇ ⁇ 2.
  • the individual filter functions can be implemented as high-pass filter functions or low-pass filter functions .
  • a bandpass filter function is calculated from the individual filter functions . This can be done in particular by subtracting two high-pass filter functions or two low-pass filter functions.
  • the provided filter function is applied to the audio signal in the time domain.
  • a fourth method step 140 the filtered audio signal is output.
  • the filtered audio signal can be output either by means of a loudspeaker or another output unit, or by signal transmission to an external device.
  • Fig. 2 shows an exemplary embodiment of the method according to the invention.
  • an input signal ⁇ ( ⁇ ) is transmitted to a plurality of high-pass filters 10 via a parallel connection, wherein Fig. 2 shows a first high-pass filter 10a, a second high-pass filter 10b and a ⁇ -th high-pass filter 10c.
  • ⁇ with ⁇ ⁇ [1 .. ⁇ ] has the cutoff frequencies ⁇ , ⁇ , where ⁇ , ⁇ ⁇ ⁇ , ⁇ .
  • the specific design of the cutoff frequencies can vary depending on the application scenario.
  • the high-pass filtered signals ⁇ ( ⁇ ) are output at the high-pass filter 10. Neighboring high-pass filtered signals are then subtracted from each other to produce bandpass signals. March 14, 2025
  • the bandpass signals are then weighted with time-variant weighting factors ⁇ ( ⁇ ) and then combined to form the output signal summed up.
  • the weighting factors ⁇ ( ⁇ ) can preferably assume a value from 0 to 1. In some embodiments of the present invention, it can also be provided that the weighting factors can also assume values that are greater than 1 or less than 0.
  • the weighting factors are also calculated depending on the specific application. When calculating the weighting factors, the approaches known from the prior art can be used.
  • the individual high-pass filters In order to generate corresponding bandpass signals from the individual high-pass filters by subtraction, it is preferred that the individual high-pass filters have certain magnitude and phase properties. These preferred properties are shown by way of example in Figs. 3 and 4. In particular, it can be seen in Fig. 3 that the magnitude responses 20 for the individual high-pass filters approach each other at high frequencies. Fig. 3 shows a first magnitude response 20a, a second magnitude response 20b, a third magnitude response 20c, and a fourth magnitude response 20d. As can be seen in Fig. 3 , the magnitude responses 20 are closely spaced from one another above a certain frequency (in particular above an upper stopband frequency ⁇ , ⁇ ) . Furthermore, Fig. 4 shows that the phase responses 20 also converge at high frequencies. Fig.
  • the neighboring high-pass filters can , in particular, be designed such that their magnitude and/or phase response approach each other above an upper stopband frequency ⁇ , ⁇ , so that subtraction leads to destructive interference.
  • the cutoff frequencies of the high-pass filters, and accordingly the passbands of the band-pass filters can advantageously be distributed evenly on a psychoacoustically motivated frequency scale, such as the Bark scale. This allows the input signal to be processed in frequency bands that mimic the human ear. However, the choice of cutoff frequencies depends on the specific application scenario. The present invention is not limited to a specific choice of cutoff frequencies.
  • the overall transfer function is therefore defined exclusively by the first high-pass filter. This makes the overall transfer function smooth in the passband and has a short group delay.
  • the high-pass filters can be designed, for example, using an optimization method with a cost function based on the magnitude response of the bandpass filters in the passband and stopband, as well as the sum of all bandpass filters .
  • the filters can be implemented as FIR or IIR filters. They can advantageously be minimum-phase, exhibiting a nonlinear phase response.
  • a filter can be described as minimum-phase if its zeros, i.e., the zeros of the numerator polynomial of its filter transfer function, lie within the unit circle or have an amplitude ⁇ 1.
  • This definition applies to FIR filters as well as to IIR filters that are not implemented as all-pole filters, i.e., filters whose transfer function comprises only denominator coefficients and, if applicable, a gain factor. All-pole filters are, by definition, minimum-phase.
  • the high-pass filters can also advantageously be optimized so that the overall transfer function ⁇ ( ⁇ ) follows a desired magnitude and phase response, so that the filter bank, for example, implicitly performs frequency weighting or equalization.
  • Fig. 5 shows the resulting magnitude responses 22 of the exemplary bandpass filters based on the high-pass filters shown in Figs. 3 and 4.
  • Fig. 5 shows a first magnitude response 22a, a second magnitude response 22b, a third magnitude response 22c, and a fourth magnitude response 22d for the corresponding bandpass filters.
  • the overall transfer function ⁇ ( ⁇ ) then corresponds to the high-pass filter with the solid line from Fig. 3 and Fig. 4.
  • low-pass filters 11 also be used to implement the inventive method, as shown by way of example in Fig. 6.
  • Fig. 6 shows the resulting magnitude responses 22 of the exemplary bandpass filters based on the high-pass filters shown in Figs. 3 and 4.
  • Fig. 5 shows a first magnitude response 22a, a second magnitude response 22b, a third magnitude response 22c, and a fourth magnitude response 22d for the corresponding bandpass filters.
  • the overall transfer function ⁇ ( ⁇ ) then corresponds to the high-pass filter with the solid line from Fig
  • a first low-pass filter 11a, a second low-pass filter 11b and a ⁇ -th March 14, 2025 Low-pass filter 11c is shown.
  • the basic principle of bandpass behavior described above based on the characteristics of the filters as well as destructive interference by subtraction still applies, but some adjustments have to be made compared to the embodiment shown in Fig. 2.
  • Fig. 7 shows a further embodiment of the method according to the invention, wherein, compared to the embodiment shown in Fig. 2, the explicit difference formation of the high-pass signals ⁇ ( ⁇ ) has been removed.

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  • Physics & Mathematics (AREA)
  • Engineering & Computer Science (AREA)
  • Acoustics & Sound (AREA)
  • Signal Processing (AREA)
  • Networks Using Active Elements (AREA)
  • Tone Control, Compression And Expansion, Limiting Amplitude (AREA)
  • Filters That Use Time-Delay Elements (AREA)

Abstract

L'invention concerne un procédé (100) utilisé pour traiter des signaux audio, comprenant les étapes suivantes : fournir (110) un signal audio x(ռ) ; fournir (120) une fonction de filtre ; appliquer (130) la fonction de filtre au signal audio x(ռ) dans le domaine temporel de façon à obtenir un signal audio filtré x(ռ) ; délivrer en sortie (140) le signal audio filtré x(ռ) ; la fonction de filtre étant fournie sur la base de K fonctions de filtre individuelles, K ≥ 2 ; et les fonctions de filtre individuelles comprenant au moins deux fonctions de filtre passe-haut ou au moins deux fonctions de filtre passe-bas à partir desquelles une fonction de filtre passe-bande est déterminée.
PCT/EP2025/057939 2024-03-22 2025-03-24 Procédé de traitement de signaux audio Pending WO2025196331A1 (fr)

Applications Claiming Priority (2)

Application Number Priority Date Filing Date Title
DE102024108243.7 2024-03-22
DE102024108243.7A DE102024108243B4 (de) 2024-03-22 2024-03-22 Verfahren zur Verarbeitung von Audiosignalen

Publications (1)

Publication Number Publication Date
WO2025196331A1 true WO2025196331A1 (fr) 2025-09-25

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Family Applications (1)

Application Number Title Priority Date Filing Date
PCT/EP2025/057939 Pending WO2025196331A1 (fr) 2024-03-22 2025-03-24 Procédé de traitement de signaux audio

Country Status (2)

Country Link
DE (1) DE102024108243B4 (fr)
WO (1) WO2025196331A1 (fr)

Citations (2)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US6141672A (en) 1997-09-02 2000-10-31 Temic Telefunken Microelectronic Gmbh Tunable digital filter arrangement
DE102022111300A1 (de) 2022-05-06 2023-11-09 Elevear GmbH Vorrichtung zur Reduzierung des Rauschens bei der Wiedergabe eines Audiosignals mit einem Kopfhörer oder Hörgerät und entsprechendes Verfahren

Patent Citations (2)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US6141672A (en) 1997-09-02 2000-10-31 Temic Telefunken Microelectronic Gmbh Tunable digital filter arrangement
DE102022111300A1 (de) 2022-05-06 2023-11-09 Elevear GmbH Vorrichtung zur Reduzierung des Rauschens bei der Wiedergabe eines Audiosignals mit einem Kopfhörer oder Hörgerät und entsprechendes Verfahren

Non-Patent Citations (2)

* Cited by examiner, † Cited by third party
Title
JAMES L CROWLEY: "A Representation for Visual Information", 1 November 1981 (1981-11-01), Pittsburgh, Pennsylvania, USA, XP055186471, Retrieved from the Internet <URL:http://www-prima.imag.fr/jlc/papers/Crowley-Thesis81.pdf> [retrieved on 20150428] *
SHARMA D P ET AL: "Design and implementation of emission filter for soft modem on TMS320C50 DSP chip", CIRCUITS AND SYSTEMS, 2002. APCCAS '02. 2002 ASIA-PACIFIC CONFERENCE O N OCT. 28-31, 2002, PISCATAWAY, NJ, USA,IEEE, vol. 2, 28 October 2002 (2002-10-28), pages 499 - 503, XP010620872, ISBN: 978-0-7803-7690-8 *

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DE102024108243A1 (de) 2025-09-25
DE102024108243B4 (de) 2025-10-02

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