EP2257082A1 - Hintergrundgeräusch-Schätzung in einem Lautsprecher-Raum-Mikrophon-System - Google Patents

Hintergrundgeräusch-Schätzung in einem Lautsprecher-Raum-Mikrophon-System Download PDF

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Publication number
EP2257082A1
EP2257082A1 EP09161443A EP09161443A EP2257082A1 EP 2257082 A1 EP2257082 A1 EP 2257082A1 EP 09161443 A EP09161443 A EP 09161443A EP 09161443 A EP09161443 A EP 09161443A EP 2257082 A1 EP2257082 A1 EP 2257082A1
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Prior art keywords
signal
microphone
loudspeakers
lrm
set forth
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EP09161443A
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English (en)
French (fr)
Inventor
Markus Christoph
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Harman Becker Automotive Systems GmbH
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Harman Becker Automotive Systems GmbH
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Priority to EP09161443A priority Critical patent/EP2257082A1/de
Publication of EP2257082A1 publication Critical patent/EP2257082A1/de
Withdrawn legal-status Critical Current

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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
    • H04R27/00—Public address systems
    • 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
    • G10K2210/3055—Transfer function of the acoustic system
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04S—STEREOPHONIC SYSTEMS 
    • H04S3/00—Systems employing more than two channels, e.g. quadraphonic
    • H04S3/02—Systems employing more than two channels, e.g. quadraphonic of the matrix type, i.e. in which input signals are combined algebraically, e.g. after having been phase shifted with respect to each other

Definitions

  • the invention relates to an assembly for multichannel echo cancellation, particularly to an assembly for multichannel cancellation of the music signal component of a microphone signal.
  • background noise in this context generally includes both acoustic waves acting from outside such as, for example, environmental noise or the noise of the vehicle on the move as picked up in the passenger compartment of a motor vehicle, as well as acoustic waves triggered by vibration, for instance of the passenger compartment or transmission of a motor vehicle. When this noise is unwanted it is also termed nuisance noise.
  • an electro-acoustic (audio) system in a noisy environment such as, for instance the passenger compartment of a motor vehicle, this may likewise be a nuisance to voice communication.
  • This background noise may involve noise stemming from the wind, the engine of the motor vehicle, the tyres, a blower and other components of the motor vehicle and is thus a function of the speed, tyre/road contact and operating conditions of the motor vehicle on the move.
  • an assembly in which the signal components of a plurality of non-interdependent acoustic channels of a multichannel music source are processed in a multichannel assembly for acoustic echo cancellation (AEC) so that the music components in the signal of a microphone are optimally cancelled.
  • AEC acoustic echo cancellation
  • the microphone is located in a closed-off acoustic room, such as, for example, the passenger compartment of a motor vehicle and picks up signal components of music, background noise and any speech signals existing.
  • adaptive filters are used in recursive methods to approximate a wanted impulse response or the transfer function of an unknown system with sufficient accuracy, as is also termed estimating the transfer function of an unknown system.
  • Adaptive filters are understood to be digital filters realized typically with the aid of algorithms on digital signal processors and which adapt their filter coefficients by a given algorithm to an input signal.
  • an unknown system is assumed to be a linear distorting system whose transfer function is sought.
  • an adaptive system is circuited in parallel with the unknown system.
  • Adaptive processes have the advantage that by continually changing the filter coefficients the algorithms automatically adapt also to changing conditions of the surroundings, for example, room changes due to different passenger and luggage situations within the passenger compartment.
  • This capability is achieved by a recursive system structure which continually optimizes the parameters.
  • the unknown system may be, for example, the passenger compartment of a motor vehicle in which a signal (for example speech and/or music) projected by one or more loudspeakers, is filtered via the unknown transfer function of the room and picked up by a microphone in this room.
  • the basic principle of an adaptive filter realized in a digital signal processor is shown in FIG. 1 .
  • the arrangement of FIG. 1 comprises an "unknown system” U and an "adaptive filter” A.
  • the unknown system U may be, for example the unknown acoustic transfer function of the passenger compartment of a motor vehicle.
  • an input signal x(n) is converted by the unknown system U, for example an acoustic transfer path into a signal d(n).
  • the input signal x(n) is translated by the adaptive filter A into the signal y(n) .
  • the signal d(n) distorted by the transfer function of the unknown system U serves as the desired reference signal from which the output y(n) of the adaptive filter A is deducted in thus generating an error signal e(n).
  • the filter coefficients are set by iteration so that the error signal e(n) is minimized, resulting in y(n) approximating d(n). This achieves approximation of the unknown system U and thus also of its transfer function in maximizing extinction of the signal d(n) by the signal y(n).
  • the LMS algorithm is an algorithm for approximating the solution of the known LMS problem as often occurs, for example, in application of adaptive filters realized in digital signal processors.
  • the algorithm is based on the so-called method of steepest descent (gradient descent method) for simple approximation of the gradient.
  • the algorithm works recursive in time, i.e. with every new set of data the algorithm is reactivated and the solution updated. Because of its low complexity, numerical stability and low memory requirement the LMS algorithm is often employed for adaptive filters and adaptive controls. Further methods could be, for example, recursive least squares, QR decomposition least squares, least squares lattice, zero-forcing, stochastic gradient methods, etc.
  • Adaptive filters include infinite impulse response (IIR) filters or finite impulse response (FIT) filters.
  • FIR filters are characterized by having a finite impulse response and working in discrete steps in time as are usually determined by the sampling frequency of an analog signal.
  • y(n) is the starting value at the point in time n as computed from the sum of the N last sampled input values x(n-N-1) to x(n) weighted with the filter coefficients b i .
  • the transfer function to be approximated is realized as described above, for example, by modifying these
  • IIR filters Unlike FIR filters, IIR filters also take into account the already computed starting values (recursive filter) and are characterized by having an infinite impulse response. But since the computed values are very small after a finite time, computation can be discontinued after a finite number of sampling values n in actual practice.
  • y(n) is the starting value at the point in time n as computed from the sum of the sampled input values x(n) weighted with the filter coefficients b i added to the sum of the starting values y(n) weighted with the filter coefficients a i .
  • Specifying the filter coefficients a i and b i realizes in turn the wanted transfer function.
  • IIR filters may be unstable, but attain higher selectivity for the same expense in realization.
  • the filter that is selected is the filter which best satisfies the necessary conditions, taking into account the requirements and the complexity of the computation involved.
  • FIG. 2 there is illustrated diagrammatically the sequence of a typical LMS algorithm for iterative adaptation of a FIR filter by way of example.
  • FIG. 2 includes the reference signal x(n) already known from FIG. 1 as a first input signal for the adaptive LMS algorithm as well as, as a second input signal, the signal d(n) resulting from x(n) distorted as shown in FIG. 1 from the unknown system by the transfer function thereof.
  • these input signals may be sound signals converted by microphones into electrical signals, but just as well may include electrical signals, generated for example by sensors for picking up mechanical oscillations or also by tachometers, for example in means for reducing the noise in motor vehicles.
  • FIG. 2 Furthermore included in FIG. 2 is the diagrammatic representation of an n th order filter (depicted here in the embodiment of a FIR filter by way of example) by which a reference signal x(n) is converted into a signal y(n) at the point in time n.
  • the filter coefficients of the adaptive filter as are actual at the point in time n as shown in FIG. 2 are identified b 0 (n), b 1 (n) ... b N-1 (n).
  • the adaptation algorithm changes the filter parameters iteratively until the error signal or difference signal e(n) between the signal d(n), distorted by the transfer function of the unknown system, and the filtered reference signal y(n)is minimized.
  • the two input signals x(n) and d(n) are generally stochastic signals, in the case of acoustic echo cancellation (AEC) systems, for example, noisy detected signals, activation signals or communication signals.
  • AEC acoustic echo cancellation
  • the quality factor as expressed by the MSE can be minimized by a simple recursive algorithm, the said least mean square (LMS) algorithm.
  • LMS least mean square
  • the function to be minimized is the square of the error, meaning that for a better approximation of the minimum of the error squared, simply the error itself, multiplied by a constant, is added to the approximation as last found before.
  • the adaptive FIR filter must be selected at least as long as the relevant component of the unknown impulse response of the unknown system to be approximated so that the adaptive filter has a sufficient degree of freedom to really minimize the error signal e(n).
  • the filter coefficients are changed stepwise in the direction of the maximum reduction or negative gradient of the degree of error MSE, the parameter ⁇ regulating the step size.
  • the new filter coefficients b k (n+1) correspond to the old filter coefficients b k (n) plus a correctional term which depends on the error signal e(n) (see FIG. 1 ) and the value x(n-k) assigned to each filter coefficient b k .
  • the LMS convergence parameter ⁇ also referred to as gain factor or step size, represents a measure for how fast and stable the filter is adapted.
  • FIG. 3 there is illustrated, by way of example, a loudspeaker room microphone (LRM) system employing an adaptive filter for echo cancellation, it shows a loudspeaker L, a signal source S and a microphone M. Also shown in FIG. 3 is a signal x(n) and the impulse response h(n) of the transfer path between the loudspeaker L and the microphone M.
  • FIG. 3 shows the principle structure of a signal processing path for cancellation of echo signals, this signal processing path comprising an adaptive filter ⁇ ( n ) and a summing block ⁇ 1 .
  • FIG. 3 shows the principle structure of a signal processing path for cancellation of echo signals, this signal processing path comprising an adaptive filter ⁇ ( n ) and a summing block ⁇ 1 .
  • FIG. 4 there is illustrated an assembly of an acoustic stereo echo compensator which uses two non-interdependent signals of a stereo signal source for echo compensation.
  • FIG. 4 shows a stereo signal source S, a loudspeaker room microphone (LRM) system, two loudspeakers L1 and L2, a microphone M, two summing units ⁇ 1 and ⁇ 2 as well as two adaptive filters A1 and A2.
  • the stereo signal source S generates two non-interdependent signals X 1 (z) and X 2 (z) serving as input signals for the loudspeakers L1 and L2 arranged in the loudspeaker room microphone (LRM) system.
  • the microphone M Likewise arranged in the loudspeaker room microphone (LRM) system is the microphone M.
  • the signal X 1 (z) serves as the input signal for the adaptive filter A1 and the signal X 2 (z) serves as the input signal for the adaptive filter A2.
  • the adaptive filters A1 and A2 are used to approximate the transfer functions H 1 (z) and H 2 (z), for example, by using an NLMS algorithm.
  • the output signals Y 1 (z) and Y 2 (z) of the adaptive filters A1 and A2 are added using the summing unit ⁇ 1 and the negated result Y(z) is added to the output signal D(z) of the microphone M using the summing unit ⁇ 2.
  • the microphone signal D(z) of the microphone M includes, in addition to the music signal components of the loudspeakers L1 and L2, also any background noise (for example, the noise of the vehicle on the move) existing and any speech signal components of the vehicle occupants.
  • H ⁇ l , k H ⁇ 1 ⁇ l , k T , H ⁇ 2 ⁇ l , k T T , ⁇ is the adaptation step size, and S
  • an assembly for suppressing music signal components in a microphone signal may also comprise a plurality of adaptive filters for approximating all or some of the plurality of transfer functions.
  • FIG. 5 there is illustrated in a block circuit diagram a prior art assembly of a multichannel active matrix decoding system for stereo input signals, known as a logic7 ® decoding system.
  • FIG. 5 shows a logic7 ® matrix decoder 1, eight signal amplifier units 2, 3, 4, 5, 6, 7, 8 and 9 and eight loudspeakers 10, 11, 12, 13, 14, 15, 16 and 17.
  • the logic7 ® matrix decoder 1 comprises two signal inputs 18 and 19 for stereo input signals 20 and 21, the signal input 18 serving to receive the stereo input signal 20 of the left channel of a dual channel stereo signal and signal input 19 serving to receive the stereo input signal 21 of the right channel of a dual channel stereo signal.
  • the logic7 ® matrix decoder 1 comprises furthermore eight signal outputs for the signals 22, 23, 24, 25, 26, 27, 28 and 29.
  • the logic7 ® matrix decoder 1 generates from stereo input signals 20 (left stereo channel) and 21 (right stereo channel) non-interdependent signals 22, 23, 24, 25, 26, 27, 28 and 29 which are amplified by corresponding output signal amplifier units 2, 3, 4, 5, 6, 7, 8 and 9 and forwarded to corresponding loudspeakers 10, 11, 12, 13, 14, 15, 16 and 17 of a multichannel audio system.
  • the amplified signal 22 serves to activate the loudspeaker 10 which in a given room corresponds to a left front loudspeaker situated front, left relative to the position of a listener.
  • the amplified signal 24 serves to activate the loudspeaker 12 which in a given room corresponds to a right front loudspeaker situated front, right relative to the position of a listener.
  • the amplified signal 23 serves to activate the loudspeaker 11 which in a given room corresponds to a center loudspeaker situated between the left front loudspeaker 10 and the right front loudspeaker 12.
  • the amplified signals 25, 26, 27, 28 and 29 correspondingly serve to activate the loudspeakers 13, 14, 15, 16 and 17.
  • the loudspeaker 13 is positioned left relative to a listener and loudspeaker 14 positioned right relative to a listener whilst loudspeaker 15 is positioned rear left relative to a listener and loudspeaker 16 is positioned rear right relative to a listener.
  • the signal 29 amplified by the signal amplifier unit 9 serves to activate the subwoofer 17 which exclusively serves to replicate the low-frequency signal components of the audio signal, it making no contribution to the surround effect of replication created by the loudspeakers 10, 11, 12, 13, 14, 15 and 16.
  • FIG. 6 there is illustrated a block circuit diagram of an assembly of a seven-channel audio system as shown in FIG. 5 in a given room which may be, for example, the passenger compartment of a motor vehicle.
  • FIG. 6 shows a left front loudspeaker 10, a right front loudspeaker 12, a loudspeaker 11 positioned in the middle between the left and right front loudspeakers, a side loudspeaker 13 positioned left, a right side loudspeaker 14, a left rear loudspeaker 15 and a right rear loudspeaker 16.
  • the subwoofer 17 likewise usually included in a seven-channel audio system.
  • FIG. 6 Apart furthermore from FIG. 6 are six signal processing blocks 32, 33, 34, 35, 36 and 37 as components of the logic7 ® matrix decoder 1 as shown in FIG. 5 to generate the corresponding signals 22, 23, 24, 25, 26, 27, 28 and 29 for activating the loudspeakers 10, 11, 12, 13, 14, 15, 16 and 17 (see FIG. 5 ).
  • Signal components for the left front loudspeaker 10 and for the right front loudspeaker 12 are used in the logic7 ® matrix decoder 1 to generate therefrom the signal for the center loudspeaker 11 (as detailed below).
  • the signal processing blocks 32 and 33 serve to attenuate the amplitude of these signal components as a function of their spectral distribution and as a function of the wanted surround effect.
  • the attenuation in a logic7 ® matrix decoder 1 is usually in the range 0 dB to -7.5 dB.
  • the signal processing blocks 34, 35, 36 and 37 serve to delay in time the signals generated from both stereo input signals (see signals 20 and 21 as shown in FIG. 5 ) for the loudspeakers 13, 14, 15 and 16 for the wanted surround reverberation effect) and to shelve the level in certain frequency bands (surround effect) as is usually done by using roll-off and shelving filters.
  • Such a surround system features an adjustable time delay between the sound signals replicated by the left front loudspeaker 10 and by the side left loudspeaker 13 also termed surround loudspeaker.
  • This time delay is effected by the signal processing block 34 and amounts to roughly 8 ms as is usual for motor vehicle sound system applications, this likewise applying to the time delay between the right front loudspeaker 12 and the side right surround loudspeaker 14 as effected by the signal processing block 35.
  • such a surround system features a further adjustable time delay between the sound signals replicated by the side left loudspeaker 13 and by the rear left loudspeaker 15.
  • This time delay is effected by the signal processing block 36 and amounts to roughly 14 ms as is usual for motor vehicle sound system applications, this likewise applying to the time delay between the side right loudspeaker 14 and the rear right loudspeaker 16 as effected by the signal processing block 37.
  • an optional subwoofer as may likewise be included.
  • the object of a matrix decoder such as for example the logic7 ® matrix decoder 1 as shown in FIG. 5 is to convert signals from, for example, two input channels (stereo signals) into, for example, 7 output channels to create the wanted stereo surround effect in a given room. These output channels are used to activate loudspeakers located in various positions in the room (see FIG.
  • the signals intended to come acoustically from a certain direction are processed in the matrix decoder so that they appear to come from the corresponding direction for the listener when replicated by the loudspeakers of the audio system, thus defining for a certain point in time what is called a listener event direction and, where necessary, the location of such a listener event, both of which can change with time in a dynamic audio signal.
  • the output signals of a matrix decoder are linear combinations of two input signals (stereo signal).
  • the coefficients of the linear combinations are functions of time which change non-linearly but slowly as compared to the audible frequencies. These matrix elements may also be complex functions of the frequency and time. It is thus the task of such a decoder to define or control the response of these coefficients.
  • the simplest matrix decoder is a passive matrix decoder in which all coefficients are fixed values wherein the output signal for a left loudspeaker results from the input signal for the left channel multiplied by 1, the output signal for a center loudspeaker resulting from the input signal for the left channel multiplied by 0.7 plus the input signal for the right channel multiplied by 0.7 and the output signal for a right loudspeaker resulting from the input signal for the right channel multiplied by 1.
  • an active matrix decoder such as for example the logic7 ® matrix decoder
  • the demands on an active matrix decoder are significantly more far-reaching, also influencing the signal generated for the center loudspeaker.
  • a strongly directed signal for instance a signal component intended to be output substantially in the left portion of the replication room by a surround system
  • a strongly directed signal is a component of the stereo input signal.
  • the channels replicating no directed signal component are required to comprise only a minimum output signal.
  • a signal required to appear in the middle between a right loudspeaker and a center loudspeaker in stereo output is required not to generate any output signals for the left and rear loudspeaker of a multichannel audio system.
  • a signal intended to be output in the middle is required not to comprise signal components for left and right loudspeakers.
  • the total output signal of the decoder is required to create the same impression for the volume when the motion of a directed signal is in various portions of the surround.
  • the total energy of the non-directed signal component of an audio signal must be maintained constant in every output channel to replicate a directed signal changing in direction.
  • the transition between replication of signal components all non-directed and signal components all directed must be even with no shifts in the perceived direction of the sound rendering. All of these requirements are satisfied by the logic7 ® matrix decoder and the signals for the corresponding loudspeakers, such as the center loudspeaker of a surround system are conditioned accordingly.
  • assemblies handling multichannel non-interdependent music signals are, for example, replication systems providing discrete signals for activating loudspeakers in accordance with "Dolby Digital 5.1 ® " or "DTS 6.1 discrete” operating 6 and 7 loudspeakers respectively in an LRM system of non-interdependent activation signals.
  • FIG. 7 there is illustrated an assembly of an acoustic echo compensator (AEC) using, for example, four non-interdependent signals of a signal source for echo compensation which could also use more than four signals, however, since the signal source comprises seven non-interdependent output signals as is the case, for instance with a logic7 ® decoder.
  • AEC acoustic echo compensator
  • FIG. 7 illustrates an assembly of an acoustic echo compensator (AEC) using, for example, four non-interdependent signals of a signal source for echo compensation which could also use more than four signals, however, since the signal source comprises seven non-interdependent output signals as is the case, for instance with a logic7 ® decoder.
  • a signal source S a loudspeaker room microphone (LRM) system, seven loudspeakers L1, L2, L3, L4, L5, L6 and L7 arranged in the loudspeaker room microphone(LRM) system, a microphone 7, two summing units ⁇ 1 and ⁇ 2 as well as four
  • the signal source S generates seven non-interdependent signals X 1 (z), X 2 (z), X 3 (z), X 4 (z), X 5 (z), X 6 (z) and X 7 (z) serving as input signals for the loudspeakers L1, L2, L3, L4, L5, L6, and L7 arranged in the loudspeaker room microphone (LRM) system, also featuring a microphone M.
  • H 1 (z), H 2 (z), H 3 (z), H 4 (z), H 5 (z), H 6 (z) and H 7 (z) depend, in turn, on the actual configuration of the LRM system, such as, for example, its geometry, any reflecting surfaces and their response and any furniture or seating arrangements (in a motor vehicle passenger compartment, for instance).
  • the signal X 1 (z), X 2 (z), X 3 (z) and X 4 (z) respectively serves as the input signal for the adaptive filter A1, A2, A3 and A4.
  • the adaptive filters A1, A2, A3 and A4 the transfer functions H1(z), H 2 (z), H 3 (z) and H 4 (z) are approximated, for example in turn by using an NLMS algorithm.
  • the output signals Y 1 (z), Y 2 (z), Y 3 (z) and Y 4 (z) of the adaptive filters A1, A2, A3 and A4 are added using the summing unit ⁇ 1 and the negated result Y(z) is added to the output signal D(z) of the microphone M using the summing unit ⁇ 2.
  • the error signal E(z) which is used in turn to optimize the filter parameters of the adaptive filters A1, A2, A3 and A4 in the sense of minimizing the error signal E(z).
  • the microphone M When, for instance, the microphone M is arranged in the passenger compartment of a motor vehicle the microphone signal D(z) of the microphone M picks up, in addition to the music signal components of the loudspeakers L1, L2, L3, L4, L5, L6 and L7, also the existing background noise (for example the noise of the vehicle on the move) and any speech signal components of the vehicle occupants.
  • An assembly is shown in FIG. 7 in which four transfer functions are approximated and the corresponding signal components in the microphone signal D(z) cancelled, although, of course, further existing transfer functions may be included in the processing.
  • the formula for computing the error signal is the same as that described in conjunction with FIG 4 .
  • H ⁇ ⁇ l , k + 1 H ⁇ l ⁇ k + ⁇ * S - 1 l ⁇ k * X H l ⁇ k * E ⁇ b l ⁇ k resulting in, for the filters A1 ...
  • H ⁇ 1 ⁇ l , k + 1 H ⁇ 1 l ⁇ k + ⁇ * S - 1 l ⁇ k * X 1 H l ⁇ k * E ⁇ l ⁇ k .
  • H ⁇ n ⁇ l , k + 1 H ⁇ n l ⁇ k + ⁇ * S - 1 l ⁇ k * X n H l ⁇ k * E ⁇ l ⁇ k
  • n 4 for the assembly as exemplarily shown in FIG. 7 .
  • Each loudspeaker may be substituted by a group of loudspeakers such as, e.g., a tweeter, a midrange speaker and a woofer connected together e.g. by a passive or active filter network.
  • loudspeakers such as, e.g., a tweeter, a midrange speaker and a woofer connected together e.g. by a passive or active filter network.
  • the assembly acoustically replicates the non-interdependent activation signals of the multichannel music signals as music signals via the dedicated loudspeakers arranged in the LRM system, and converts via the microphone arranged in the LRM system the total sound level existing in situ at the microphone into a corresponding microphone signal. Furthermore, the assembly approximates the acoustic transfer functions configured in the LRM system between the loudspeakers and the microphone for each of the activation signals by each of the dedicated adaptive filter units, and filters the activation signals of the multichannel music signal source assigned to the corresponding acoustic transfer functions with the corresponding adaptive filter units approximating each of the acoustic transfer functions.

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  • Physics & Mathematics (AREA)
  • Engineering & Computer Science (AREA)
  • Acoustics & Sound (AREA)
  • Signal Processing (AREA)
  • Fittings On The Vehicle Exterior For Carrying Loads, And Devices For Holding Or Mounting Articles (AREA)
EP09161443A 2009-05-28 2009-05-28 Hintergrundgeräusch-Schätzung in einem Lautsprecher-Raum-Mikrophon-System Withdrawn EP2257082A1 (de)

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Cited By (2)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
WO2018200403A3 (en) * 2017-04-24 2019-04-04 Cirrus Logic International Semiconductor Ltd. ADAPTIVE NOISE SUPPRESSION SYSTEM BASED ON SOFTWARE RADIO
CN114175606A (zh) * 2019-06-17 2022-03-11 伯斯有限公司 模块化回波消除单元

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Publication number Priority date Publication date Assignee Title
WO2003061344A2 (en) * 2002-01-17 2003-07-24 Koninklijke Philips Electronics N.V. Multichannel echo canceller system using active audio matrix coefficients
EP1475781A2 (de) * 2003-05-02 2004-11-10 Alpine Electronics, Inc. Verfahren und Vorrichtung zur Spracherkennung

Patent Citations (2)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
WO2003061344A2 (en) * 2002-01-17 2003-07-24 Koninklijke Philips Electronics N.V. Multichannel echo canceller system using active audio matrix coefficients
EP1475781A2 (de) * 2003-05-02 2004-11-10 Alpine Electronics, Inc. Verfahren und Vorrichtung zur Spracherkennung

Cited By (5)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
WO2018200403A3 (en) * 2017-04-24 2019-04-04 Cirrus Logic International Semiconductor Ltd. ADAPTIVE NOISE SUPPRESSION SYSTEM BASED ON SOFTWARE RADIO
US10720138B2 (en) 2017-04-24 2020-07-21 Cirrus Logic, Inc. SDR-based adaptive noise cancellation (ANC) system
US11631390B2 (en) 2017-04-24 2023-04-18 Cirrus Logic, Inc. SDR-based adaptive noise cancellation (ANC) system
CN114175606A (zh) * 2019-06-17 2022-03-11 伯斯有限公司 模块化回波消除单元
CN114175606B (zh) * 2019-06-17 2024-02-06 伯斯有限公司 模块化回波消除单元

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