WO2005112849A2 - Tuned feedforward lms filter with feedback control - Google Patents

Tuned feedforward lms filter with feedback control Download PDF

Info

Publication number
WO2005112849A2
WO2005112849A2 PCT/US2005/012598 US2005012598W WO2005112849A2 WO 2005112849 A2 WO2005112849 A2 WO 2005112849A2 US 2005012598 W US2005012598 W US 2005012598W WO 2005112849 A2 WO2005112849 A2 WO 2005112849A2
Authority
WO
WIPO (PCT)
Prior art keywords
noise
lms
signal
feedforward
tuning
Prior art date
Application number
PCT/US2005/012598
Other languages
French (fr)
Other versions
WO2005112849A3 (en
Inventor
Laura R. Ray
Alexander Streeter
Original Assignee
Trustees Of Dartmouth College
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 Trustees Of Dartmouth College filed Critical Trustees Of Dartmouth College
Priority to EP05758737A priority Critical patent/EP1744713A4/en
Priority to JP2007513154A priority patent/JP2007536877A/en
Publication of WO2005112849A2 publication Critical patent/WO2005112849A2/en
Publication of WO2005112849A3 publication Critical patent/WO2005112849A3/en

Links

Classifications

    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04RLOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R1/00Details of transducers, loudspeakers or microphones
    • H04R1/10Earpieces; Attachments therefor ; Earphones; Monophonic headphones
    • H04R1/1008Earpieces of the supra-aural or circum-aural type
    • HELECTRICITY
    • H03ELECTRONIC CIRCUITRY
    • H03BGENERATION OF OSCILLATIONS, DIRECTLY OR BY FREQUENCY-CHANGING, BY CIRCUITS EMPLOYING ACTIVE ELEMENTS WHICH OPERATE IN A NON-SWITCHING MANNER; GENERATION OF NOISE BY SUCH CIRCUITS
    • H03B29/00Generation of noise currents and voltages
    • 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
    • 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
    • G10K11/17861Methods, e.g. algorithms; Devices using additional means for damping sound, e.g. using sound absorbing panels
    • 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
    • H04R1/00Details of transducers, loudspeakers or microphones
    • H04R1/10Earpieces; Attachments therefor ; Earphones; Monophonic headphones
    • H04R1/1083Reduction of ambient noise
    • 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
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04RLOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R2420/00Details of connection covered by H04R, not provided for in its groups
    • H04R2420/01Input selection or mixing for amplifiers or loudspeakers

Definitions

  • Noise cancellation systems are used in various applications ranging from telephony to acoustic noise cancellation in communication headsets. There are, however, significant difficulties in implementing such stable, high performance noise cancellation systems.
  • the well-known LMS algorithm is used to perform the noise cancellation.
  • This algorithm lacks stability in the presence of inadequate excitation, non-stationary noise fields, low signal-to-noise ratio, or finite precision effects due to numerical computations. This has resulted in many variations to the standard LMS algorithm, none of which provide satisfactory performance over a range of noise parameters.
  • the leaky LMS algorithm has received significant attention.
  • the leaky LMS algorithm first proposed by Gitlin et al . introduces a fixed leakage parameter that improves stability and robustness. However, the leakage parameter improves stability at a significant expense to noise reduction performance.
  • a feedback topology is shown in Figure 16.
  • the measured error signal e ⁇ is minimized through an infinite impulse response feedback compensator designed using traditional frequency-domain methods.
  • the feedback controller seeks to force the phase between the output signal and the error signal equal to -180 degrees for as much as the ANR frequency band as possible.
  • active noise control a high-gain control law is required to achieve this objective and to maximum ANR performance.
  • a high-gain control law leaves inadequate stability margins, and such systems destabilize easily in practice, as the transfer function of the system can vary substantially with environmental conditions.
  • ANR performance is sacrificed, thus present feedback technology exhibits narrowband performance and "spillover" or creation of noise outside of the ANR band.
  • Present commercial technology implements feedback control using analog circuitry.
  • the present invention discloses a method to automatically and adaptively tune a leaky, normalized least- mean-square (LNLMS) algorithm so as to maximize the stability and noise reduction performance in feedforward adaptive noise cancellation systems.
  • the automatic tuning method provides for time-varying tuning parameters ⁇ k and ⁇ k that are functions of the instantaneous measured acoustic noise signal, weight vector length, and measurement noise variance.
  • the method addresses situations in which signal-to-noise ratio varies substantially due to nonstationary noise fields, affecting stability, convergence, and steady-state noise cancellation performance of 'LMS algorithms.
  • the method has been embodied in the particular context of active noise cancellation in communication headsets.
  • the method is generic, in that it is applicable to a wide range of systems subject to nonstationary, i.e., time-varying, noise fields, including sonar, radar, echo cancellation, and telephony.
  • the hybridization of the disclosed Lyapunov-tuned feedforward LMS filter with a feedback controller as also disclosed herein enhances stability margins, robustness, and further enhances performance.
  • the present invention is not intended to be limited to a device or method which must satisfy one or more of any stated or implied objects or features of the invention. It is also important to note that the present invention is not limited to the preferred, exemplary, or primary embodiment (s) described herein. Modifications and substitutions by one of ordinary skill in the art are considered to be within the scope of the present invention, which is not to be limited except by the following claims .
  • FIG. 1 is block diagram of one implementation of the a system on which the method of tuning an adaptive leaky
  • FIG. 2 is schematic view of the experimental embodiment of the disclosed invention
  • FIG. 3 is a schematic view of a test cell utilized for verifying the experimental results of the present invention.
  • FIGS. 4A and 4B are graphs showing active and passive SPL attenuation for a sum of pure tones between 50 and 200 Hz as measured at a microphone mounted approximately at the location of a user's ear, and two headsets, one of which embodies the present invention;
  • FIG. 5 illustrates the weight error function projected embodiment of the present invention
  • FIGS. 6A-6I show plots of a Lyapunov function difference, V ⁇ - V ⁇ , vs. parameters A and B defined in eq. 30 and 31 for signal-to-noise ratio (SNR) of 2, 10, and 100, and a filter length of 20;
  • SNR signal-to-noise ratio
  • FIG. 7 shows numerical results corresponding to the graphs of FIG. 6; and [0016]
  • FIG. 8 is a graph of a representative power spectrum of aircraft noise for experimental evaluation of the tuned leaky LMS algorithm of the present invention showing statistically determined upper and lower bounds on the power spectrum and the band limited frequency range used in experimental testing;
  • FIG. 9 is a table showing the experimentally determined mean tuning parameters for three candidate adaptive LNLMS algorithms.
  • FIG. 10 is a graph of the performance of empirically tuned NLMS and LNLMS algorithms for nonstationary aircraft noise at 100 dB;
  • FIG. 11 is a graph of the performance of empirically tuned NLMS and LNLMS algorithms for nonstationary aircraft noise at 80 dB;
  • FIGS. 12A and 12B show RMS weight vector trajectory for empirically tuned NLMS and LNLMS algorithms for nonstationary aircraft noise at 100 dB SPL and 80 dB SPL respectively;
  • FIG. 13 is a graph of the performance of three candidate-tuned LNLMS LLMS algorithms for nonstationary aircraft noise as 100 dB in which candidate 1 represents equations 33 and 34, candidate 2 equations 33 and 37, and candidate 3 equations 38 and 43;
  • FIG. 14 is a graph of the performance of three candidate-tuned LNLMS LLMS algorithms for nonstationary aircraft noise at 80 dB in which candidate 1 represents equations 33 and 34, candidate 2 equations 33 and 37, and candidate 3 equations 38 and 43;
  • FIG. 15 is a graph showing RMS weight vector histories for both 80 dB and 100 dB SPL;
  • FIG. 16 is a schematic diagram of the prior art ANR architecture
  • FIG. 17 is a schematic diagram of combined feedforward-feedback topology in accordance with one aspect of the present invention.
  • FIG.18 is a graph illustrating the active attenuation performance of each individual system/method in response to puretone noise.
  • FIG. 19 is a graph illustrating experimentally determined maximum stable gains of the disclosed feedforward system and method with and without a feedback component.
  • FIG. 1 is an embodiment of an adaptive LMS filter 10 in the context of active noise reduction in a communication headset.
  • the external acoustic noise signal 12, X k is measured by a microphone 14.
  • the external acoustic noise signal is naturally attenuated passively 16, as it passes through damping material, for example, a headset shell structure, and is absorbed by foam liners within the ear cup of the headset, as defined on [0061] .
  • the attenuated noise signal 18 is then cancelled by an equal and opposite acoustic noise cancellation signal 20, y k i generated using a speaker 22 inside the ear cup of the communication headset.
  • the algorithm 24 that computes y k is the focus of the present invention. Termed an adaptive feedforward noise cancellation algorithm in the block diagram, it provides the cancellation signal as a function of the measured acoustic noise signal X k (14'), and the error signal e k (26) , which is a measure of the residual noise after cancellation.
  • each of these measured signals contains measurement noise due to microphones and associated electronics and digital quantization.
  • Current embodiments of the adaptive feedforward noise canceling algorithm include two parameters - an adaptive step size ⁇ k that governs convergence of the estimated noise cancellation signal, and a leakage parameter ⁇ .
  • the traditional normalized, leaky feedforward LMS algorithm is given by the following two equations:
  • W k is a weight vector, or set of coefficients of a finite-impulse response filter.
  • 1 for ideal conditions: no measurement noise; no quantization noise; deterministic and statistically stationary acoustic inputs; discrete frequency components in X k ; and infinite precision arithmetic. Under these ideal conditions, the filter coefficients converge to those required to minimize the mean-squared error e k .
  • Algorithms for selecting parameter ⁇ k appear in the literature and modifications or embodiments of published ⁇ k selection algorithms appear in various prior art.
  • the choice of parameters ⁇ and ⁇ k as presented in the prior art does not guarantee stability of the traditional LMS algorithm under non-ideal real-world conditions, in which measurement noise in the microphone signals is present, finite precision effects reduce the accuracy of numerical computations, and noise fields are highly nonstationary.
  • the leakage parameter must be selected so as to maintain stability for worst case, i.e., nonstationary noise fields with impulsive noise content, resulting in significant noise cancellation degradation.
  • the invention disclosed here is a computational method, based on a Lyapunov tuning approach, and its embodiment that automatically tunes time varying parameters ⁇ k and ⁇ k so as to maximize stability with minimal reduction in performance under noise conditions with persistent or periodic low signal-to-noise ratio, low excitation levels, and nonstationary noise fields.
  • the automatic tuning method provides for time-varying tuning parameters ⁇ k and ⁇ k that are functions of the instantaneous measured acoustic noise signal X k , weight vector length, and measurement noise variance.
  • X k + Q k is the measured reference signal, which contains measurement noise Q k due to electronic noise and quantization.
  • the measurement noise is of known variance ⁇ g 2 L is the length of weight vector W k .
  • the prototype headset consists of a shell from a commercial headset, which has been modified to include ANR hardware components, i.e., an internal error sensing microphone, a cancellation speaker, and an external reference noise sensing microphone.
  • ANR hardware components i.e., an internal error sensing microphone, a cancellation speaker, and an external reference noise sensing microphone.
  • the tuning method of the present invention is embodied as software within a commercial DSP system, the dSPACE DS 1103.
  • a block diagram 30, Fig. 2 shows one implementation of the present invention.
  • the preferred embodiment of the 'Adaptive Leaky LMS' 24 contains a c- program that embodies the tuning method of the present invention, although a software implementation is not specific to nor a limitation of the present invention, but is applicable to all feedforward adaptive noise cancellation system embodiments.
  • the three inputs to the Adaptive Leaky LMS block are the reference noise 14', the error microphone 26, and a reset' trigger 32 that is implemented for experimental analysis.
  • the output signals are the acoustic noise cancellation signal 20, the tuned parameters ⁇ k (34) and ⁇ k (36), and the filter coefficients 38.
  • ANR Active Noise Reduction
  • the B&K microphone 44 which was mounted approximately at the location of a user's ear, was used to record sound pressure level (SPL) attenuation performance.
  • SPL sound pressure level
  • E ⁇ X d f c is the cross correlation between the input vector and process output.
  • LMS has some drawbacks.
  • high input power leads to large weight updates and large excess mean-square error at convergence.
  • Operating at the largest possible step size enhances convergence, but also causes large excess mean- square error, or noise in the weight vector, at convergence.
  • a nonstationary input dictates a large adaptive step size for enhanced tracking, thus the LMS algorithm is not guaranteed to converge for nonstationary inputs.
  • the stability analysis objective is to find operating bounds on the variable leakage parameter ⁇ k and the adaptive step size ⁇ k to maintain stability in the presence of noise vector Q k whose elements have known variance, given the dynamic range or a lower bound on the signal-to-noise ratio.
  • the present invention seeks time-varying parameters ⁇ and ⁇ k such that certain stability conditions on a candidate Lyapunov function V k are satisfied for all k in the presence of quantifiable noise on reference input X k .
  • the choice of ⁇ k and ⁇ k should be dependent on measurable quantities, such that a parameter selection algorithm can be implemented in realtime.
  • the selection algorithm should be computationally efficient.
  • V M ⁇ V k (l - ⁇ )W k u k u ⁇ W k + ⁇ W:u k u ⁇ W 0 + ⁇ 2 'Wa k a ⁇ W 0 (28) + 2 ⁇ k ⁇ h W k u k u . ⁇ TTWT ⁇ 0 + 2 ⁇ k ⁇ 2k Wk 1
  • the goal of the Lyapunov analysis is to enable quantitative comparison of stability and performance tradeoffs for candidate tuning rules. Since uniform asymptotic stability suffices to make such comparisons, and since the Lyapunov function of Eq. 20 enhances the ability to make such comparisons, it was selected for the analysis that follows .
  • V k+ ⁇ — V k results only if Y ⁇ k W u k u k W 0 + ⁇ k W a k a k W 0 ⁇ -2 ⁇ lk ⁇ lk W/u k k W 0 with Y lk Y 2j > 0- That the leaky LMS algorithm, as examined using the Lyapunov candidate of Eq. 20, is biased away from W 0 is in agreement with the prior art.
  • the approach taken in the present invention is to define the region of stability around the Wiener solution in terms of parameters:
  • the parameters A and B physically represent the output error ratio between the actual output and ideal output for a system converged to the Wiener solution, and the output noise ratio, or portion of the ideal output that is due to noise vector Q k . Physically, these parameters are inherently statistically bounded based on i) the maximum output that a real system is capable of producing, ii) signal-to-noise ratio in the system, and iii) the convergence behavior of the system.
  • V k+ ⁇ - v k ( 32 )
  • an adaptive step size and/or leakage parameter that simplifies analysis of Eq. 32, one can parameterize and subsequently determine conditions on remaining scalar parameters such that V k — V k ⁇ 0 for the largest region possible around the Wiener solution.
  • Such a region is now defined by parameters A and B, providing a means to graphically display the stable region and to visualize performance/stability tradeoffs introduced for candidate leakage and step size parameters.
  • the first candidate uses a traditional choice for leakage parameter in combination with a traditional choice for adaptive step size to provide:
  • the combined candidate step size and leakage factor parameterize Eq. 32 in terms of ⁇ 0 .
  • the optimal 0 one can perform a scalar optimization of V k+ ⁇ — V k with respect to ⁇ 0 and evaluate the result for worst-case constants A and J3. In essence, one seeks the value of ⁇ 0 that makes V k . ⁇ - V k most negative for worst-case deviations of weight vector W k from the Wiener solution and for worst-case effects of measurement noise Q k .
  • Worst case A and B are chosen to be that combination in the range A m ⁇ n ⁇ A ⁇ 0 and 0 ⁇ A ⁇ A max , B m ⁇ n ⁇ B ⁇ B max that provides the smallest (i.e., most conservative) step size parameter ⁇ 0 .
  • the second candidate also retains the traditional leakage factor of Eq.
  • Equation 43 is a function of statistical and measurable quantities, and is a good approximation of Eq. 39 when ⁇
  • the corresponding definitions of ⁇ Y ⁇ k Yl k > ⁇ &Mkr Ec 3- 32 becomes
  • the three candidate adaptive leakage factor and step size solutions are Candidate 1: Eq. 33 and 34, Candidate 2: Eq. 33 and 37, and Candidate 3: Eq. 38 and 43. All are computationally efficient, requiring little additional computation over a fixed leakage, normalized LMS algorithm, and all three candidate tuning laws can be implemented based on knowledge of the measured, noise corrupted reference input, the variance of the measurement noise, and the filter length.
  • V k+ ⁇ ⁇ V k for various instantaneous signal-to-noise ratios
  • Figure 6 shows plots of V k ⁇ ⁇ - V k vs. A and B for SNR of 2, (Figs. 6A-6C) 10 (Figs. 6D-6F) , and 100 (Figs. 6G-6I) , and a filter length of 20. Numerical results corresponding to Figure 6 are shown in Figure 7.
  • Figure 6 includes the zero' plane, such that stability regions provided by the intersection of the Lyapunov difference with this plane can be visualized.
  • a tuning law providing a more negative V k ⁇ . ⁇ - V k in the stable region should provide the best performance, while the tuning law providing the largest region in which V k+ ⁇ - V ⁇ 0 provides the best stability.
  • Figure 7 records the maximum and minimum values of V k+ ⁇ - V k for the range of A and B examined, showing candidate 2 should provide the best performance (and least stability) , while candidate 3 provides the best overall stability/performance tradeoff for high SNR, followed by candidates 1 and 2.
  • leakage factor approaches one as signal-to-noise ratio increases, as expected, and candidate 2 provides the most aggressive step size, which relates to the larger gradient of V k+ ⁇ - V k and thus the best predicted performance.
  • An alternate view of V k+ ⁇ - V k as it relates to performance is to consider V k+ — V k as the rate of change of energy of the system. The faster the energy decreases, the faster convergence, and hence the better performance.
  • the three candidate Lyapunov tuned leaky LMS algorithm are evaluated and compared to i) an empirically tuned, fixed leakage parameter leaky, normalized LMS algorithms (LNLMS) , and ii) an empirically tuned normalized LMS algorithm with no leakage parameter (NLMS) .
  • the comparisons are made for a low-frequency single-source, single-point noise cancellation system in an acoustic test chamber (42, Fig. 3) designed to provide a highly controlled and repeatable acoustic environment with a flat frequency response over the range of 0 to 200 Hz for sound pressure levels up to 140 dB.
  • the system under study is a prototype communication headset earcup.
  • the earcup contains an external microphone to measure the reference signal, an internal microphone to measure the error signal, and an internal noise cancellation speaker to generate y k . Details regarding the prototype are given above in connection with Figure 3.
  • the reference noise is from an F-16, a representative high-performance aircraft that exhibits highly nonstationary characteristics and substantial impulsive noise content.
  • the noise source is band limited at 50 Hz to maintain a low level of low frequency distortion in the headset speaker and 200 Hz, the upper limit for a uniform sound field in the low frequency test cell.
  • Figure 8 shows the low frequency regime of the reference noise power spectrum along with statistically determined upper and lower bounds on the power spectrum that indicate the degree of nonstationarity of the noise source.
  • PSD power spectral density
  • the amplitude of the reference noise source is established to evaluate algorithm performance over a 20 dB dynamic range, i.e., sound pressure levels of 80 dB and 100 dB, as measured inside the earcup after passive attenuation.
  • the difference in sound pressure levels tests the ability of the tuned leaky LMS algorithms to adapt to different signal-to-noise ratios.
  • the two noise amplitudes represent signal-to-noise ratio (SNR) conditions for the reference microphone measurements of 35 dB and 55 dB, respectively.
  • SNR signal-to-noise ratio
  • analysis of V k ⁇ . ⁇ - V k of Eq. 32 for Lyapunov tuned candidates shows statistically determined bounds on B of -0.6 ⁇ B ⁇ 0.6, while for the 80 dB SPL (35 dB SNR) , statistically determined bounds on B are -3 ⁇ B ⁇ 3.
  • Figure 6 which gives the V k . ⁇ - V k surface for each candidate algorithm, shows that by lowering SNR to 35 dB, instability is possible for all three candidates, as the fixed step size is chosen for worst case conditions on B of - 1 ⁇ B ⁇ 1.
  • the 80 dB SPL noise source tests the limits of stability for the three candidate algorithms.
  • the quantization noise magnitude is 610e-6 V, based on a 16-bit round-off A/D converter with a +10 V range and one sign bit.
  • the candidate LMS algorithms are implemented experimentally using a dSPACE DS1103 DSP board. A filter length of 250 and weight update frequency of 5 kHz are used. The starting point for the noise segments used in the experiments is nearly identical for each test, so that noise samples between different tests overlap.
  • FIG. 10 shows experimental results for these three filters (NLMS, LNLMS (100), and LNLMS (80)) operating at 100 dB SPL.
  • the NLMS algorithm and the LNLMS tuned for 100 dB algorithm show similar performance, while the LNLMS algorithm tuned for 80 dB shows significant performance reduction at steady-state.
  • SNR is sufficiently high that only a small amount of leakage is required to guarantee stability, thus performance degradation due to the leakage factor is minimal.
  • the NLMS algorithm is stable after five seconds of operation, a slow weight drift occurs, such that the leakage factor is required.
  • Figure 11 shows results for the 80 dB SPL.
  • the low SNR causes weight instability in the NLMS algorithm during the five second experiment.
  • the mismatch in tuning conditions, i.e., using the LNLMS (100) algorithm under 80 dB SPL conditions also results in weight drift instability.
  • Evidence of instability of the NLMS and LNLMS (100) algorithms at 80 dB is shown in time histories of the root-mean square (RMS) weight vector in Figures 12A and 12B.
  • the results of Figures 10 through 12 demonstrate both the loss of stability when using an overly aggressive (large) fixed parameter leakage parameter and the loss of performance when a less aggressive (small) leakage parameter is required in order to retain stability over large changes in the dynamic range of the reference input signal.
  • the Lyapunov based tuning approach provides a candidate algorithm that retains stability and satisfactory performance in the presence of the nonstationary noise source over the 20 dB dynamic range, i.e., at both 80 and 100 dB SPL.
  • Figure 13 shows performance at 100 dB SPL
  • Figure 14 shows performance at 80 dB SPL.
  • 100 dB SPL Figure 13
  • all three candidate algorithms retain stability, and at steady-state, noise reduction performance of all three candidate algorithms exceeds that of empirically tuned leaky LMS algorithms. In fact, performance closely approximates that of the NLMS algorithm, which represents the best possible performance for a stable system, as it includes no performance degradation due to a leakage bias.
  • candidates 2 and 3 are unstable at 80 dB SPL, reflecting the fact that candidate algorithms do not necessarily guarantee uniform asymptotic stability when assumptions regarding bounds on measurement noise are exceeded.
  • Candidate 3, which was predicted by Lyapunov analysis to provide the best stability characteristics of the three candidates retains stability and provides a steady-state SPL attenuation exceeding that of the LNLMS (80) by 5 dB.
  • Figure 15 shows the RMS weight vector histories for both 80 dB and 100 dB reference input sound pressure levels, providing experimental evidence of stability of all three candidates at 100 dB SPL and of candidate 3 at 80 dB SPL.
  • Performance gains of Lyapunov tuned candidates over the fixed leakage parameter LMS algorithms are confirmed by the mean and variance of the leakage factor for each candidate, as shown in Fig. 9.
  • the variance of the leakage factor is larger for the 80 dB test condition that for the 100 dB condition, as expected, since the measured reference signal at 80 dB represents lower average and instantaneous signal-to-noise ratios.
  • FIG. 17 shows a hybrid feedforward-feedback ANR topology in accordance with the present invention.
  • a reference microphone 100 measures the primary source, which enters the unknown acoustic process H(z) 102, and the error signal 104 remaining after ANR is measured by a microphone 106.
  • an adaptive LMS filter provides a cancellations signal -y , 108.
  • the feedforward system can be thought of as providing a smaller error signal for the feedback controller to act on, since it models the unknown acoustic process, and thus the system can tolerate an overall increase in the feedback or feedforward controller gain without destabilizing the system.
  • a feedback controlled system as being acted upon by the feedforward controller, which because it is adaptive, performs its task whether or not the feedback control component is in place.
  • a broadband, feedback controller providing 5-10 dB of attenuation in the 40 Hz to 1600 Hz frequency band is paired with the feedforward controller, which is tuned according to one aspect of the present invention.
  • Both the feedback and feedforward components are implemented digitally. Because of this, no additional hardware components are required to add the feedback component beyond those used for the feedforward controller.
  • Figure 18 shows sample experimental results. At low frequencies ( ⁇ 100 Hz) , the feedback component provides 1- 8 dB of active attenuation, and the feedforward component, which is tuned according to method disclosed herein provides 15-27 dB of attenuation.
  • the hybrid system demonstrates overall performance that is greater than the sum of the individual components at frequencies below 80 Hz.
  • the exceptional performance of the hybrid system is achieved by pairing the feedforward controller tuned in accordance with the method disclosed herein with the traditional infinite impulse response feedback controller.
  • FIG. 19 shows measured stability margins of a hybrid controller from experimental evaluation of the system when applied to ANR in a hearing protector. Measurements were made using the low frequency acoustic test cell and digital signal processing development system described herein. Stability margin is measured by the tolerable increase in the feedforward controller gain ( Kff ) before the system shows evidence of instability with and without the feedback component in place. With the hybrid system, gain margin improves by a factor of 2 to over 1000 through the band evaluated.

Landscapes

  • Physics & Mathematics (AREA)
  • Engineering & Computer Science (AREA)
  • Acoustics & Sound (AREA)
  • Multimedia (AREA)
  • Signal Processing (AREA)
  • Health & Medical Sciences (AREA)
  • General Health & Medical Sciences (AREA)
  • Otolaryngology (AREA)
  • Soundproofing, Sound Blocking, And Sound Damping (AREA)
  • Measurement Of Velocity Or Position Using Acoustic Or Ultrasonic Waves (AREA)
  • Filters That Use Time-Delay Elements (AREA)

Abstract

A method to automatically and adaptively tune a leaky, normalized least-mean-square (LNLMS) algorithm (24) so as to maximize the stability and noise reduction performance in feedforward adaptive noise cancellation systems. The automatic tuning method provides for time-varying tuning parameters λk and µk that are functions of the instantaneous measured acoustic noise signal (12), weight vector length, and measurement noise variance. The method addresses situations in which signal-to-noise ratio varies substantially due to nonstationary noise fields, affecting stability, convergence, and steady-state noise cancellation performance of LMS algorithms. The method has been embodied in the particular context of active noise cancellation in communication headsets. However, the method is generic, in that it is applicable to a wide range of systems subject to nonstationary, i.e., time-varying, noise fields, including sonar, radar, echo cancellation, and telephony.

Description

TUNED FEEDFORWARD LMS FILTER WITH FEEDBACK CONTROL
BACKGROUND OF THE INVENTION [0001] Noise cancellation systems are used in various applications ranging from telephony to acoustic noise cancellation in communication headsets. There are, however, significant difficulties in implementing such stable, high performance noise cancellation systems.
[0002] In the majority of adaptive systems, the well-known LMS algorithm is used to perform the noise cancellation. This algorithm, however, lacks stability in the presence of inadequate excitation, non-stationary noise fields, low signal-to-noise ratio, or finite precision effects due to numerical computations. This has resulted in many variations to the standard LMS algorithm, none of which provide satisfactory performance over a range of noise parameters. [0003] Among the variations, the leaky LMS algorithm has received significant attention. The leaky LMS algorithm, first proposed by Gitlin et al . introduces a fixed leakage parameter that improves stability and robustness. However, the leakage parameter improves stability at a significant expense to noise reduction performance.
[0004] Thus, the current state-of-the-art LMS algorithms must tradeoff stability and performance through manual selection of tuning parameters, such as the leakage parameter. In such noise cancellation systems, a constant, manually selected tuning parameter cannot provide optimized stability and performance for a wide range of different types of noise sources such as deterministic, tonal noise, stationary random noise, and highly nonstationary noise with impulsive content, nor adapt to highly variable and large differences in the dynamic ranges evident in real-world noise environments. Hence, "worst case", i.e., highly variable, nonstationary noise environment scenarios must be used to select tuning parameters, resulting in substantial degradation of noise reduction performance over a full range of noise fields. [0005] Presently, commercial active noise reduction (ANR) technology uses feedback control to reduce unwanted sound. A feedback topology is shown in Figure 16. Here, the measured error signal e^is minimized through an infinite impulse response feedback compensator designed using traditional frequency-domain methods. The feedback controller seeks to force the phase between the output signal and the error signal equal to -180 degrees for as much as the ANR frequency band as possible. In active noise control, a high-gain control law is required to achieve this objective and to maximum ANR performance. However, a high-gain control law leaves inadequate stability margins, and such systems destabilize easily in practice, as the transfer function of the system can vary substantially with environmental conditions. In order to provide adequate stability margins, ANR performance is sacrificed, thus present feedback technology exhibits narrowband performance and "spillover" or creation of noise outside of the ANR band. Present commercial technology implements feedback control using analog circuitry. SUMMARY OF THE INVENTION
[0006] The present invention discloses a method to automatically and adaptively tune a leaky, normalized least- mean-square (LNLMS) algorithm so as to maximize the stability and noise reduction performance in feedforward adaptive noise cancellation systems. The automatic tuning method provides for time-varying tuning parameters λk and μk that are functions of the instantaneous measured acoustic noise signal, weight vector length, and measurement noise variance. The method addresses situations in which signal-to-noise ratio varies substantially due to nonstationary noise fields, affecting stability, convergence, and steady-state noise cancellation performance of 'LMS algorithms. The method has been embodied in the particular context of active noise cancellation in communication headsets. However, the method is generic, in that it is applicable to a wide range of systems subject to nonstationary, i.e., time-varying, noise fields, including sonar, radar, echo cancellation, and telephony. Further, the hybridization of the disclosed Lyapunov-tuned feedforward LMS filter with a feedback controller as also disclosed herein enhances stability margins, robustness, and further enhances performance. [0007] It is important to note that the present invention is not intended to be limited to a device or method which must satisfy one or more of any stated or implied objects or features of the invention. It is also important to note that the present invention is not limited to the preferred, exemplary, or primary embodiment (s) described herein. Modifications and substitutions by one of ordinary skill in the art are considered to be within the scope of the present invention, which is not to be limited except by the following claims .
BRIEF DESCRIPTION OF THE DRAWINGS [0008] These and other features and advantages of the- present invention will be better understood by reading the following detailed description, taken together with the drawings wherein: [0009] FIG. 1 is block diagram of one implementation of the a system on which the method of tuning an adaptive leaky
LMS filter in accordance with the present invention can be practiced;
[0010] FIG. 2 is schematic view of the experimental embodiment of the disclosed invention;
[0011] FIG. 3 is a schematic view of a test cell utilized for verifying the experimental results of the present invention;
[0012] FIGS. 4A and 4B are graphs showing active and passive SPL attenuation for a sum of pure tones between 50 and 200 Hz as measured at a microphone mounted approximately at the location of a user's ear, and two headsets, one of which embodies the present invention; [0013] FIG. 5 illustrates the weight error function projected embodiment of the present invention;
[0014] FIGS. 6A-6I show plots of a Lyapunov function difference, V^γ - V^ , vs. parameters A and B defined in eq. 30 and 31 for signal-to-noise ratio (SNR) of 2, 10, and 100, and a filter length of 20;
[0015] FIG. 7 shows numerical results corresponding to the graphs of FIG. 6; and [0016] FIG. 8 is a graph of a representative power spectrum of aircraft noise for experimental evaluation of the tuned leaky LMS algorithm of the present invention showing statistically determined upper and lower bounds on the power spectrum and the band limited frequency range used in experimental testing;
[0017] FIG. 9 is a table showing the experimentally determined mean tuning parameters for three candidate adaptive LNLMS algorithms;
[0018] FIG. 10 is a graph of the performance of empirically tuned NLMS and LNLMS algorithms for nonstationary aircraft noise at 100 dB;
[0019] FIG. 11 is a graph of the performance of empirically tuned NLMS and LNLMS algorithms for nonstationary aircraft noise at 80 dB; [0020] FIGS. 12A and 12B show RMS weight vector trajectory for empirically tuned NLMS and LNLMS algorithms for nonstationary aircraft noise at 100 dB SPL and 80 dB SPL respectively;
[0021] FIG. 13 is a graph of the performance of three candidate-tuned LNLMS LLMS algorithms for nonstationary aircraft noise as 100 dB in which candidate 1 represents equations 33 and 34, candidate 2 equations 33 and 37, and candidate 3 equations 38 and 43; [0022] FIG. 14 is a graph of the performance of three candidate-tuned LNLMS LLMS algorithms for nonstationary aircraft noise at 80 dB in which candidate 1 represents equations 33 and 34, candidate 2 equations 33 and 37, and candidate 3 equations 38 and 43;
[0023] FIG. 15 is a graph showing RMS weight vector histories for both 80 dB and 100 dB SPL;
[0024] FIG. 16 is a schematic diagram of the prior art ANR architecture; [0025] FIG. 17 is a schematic diagram of combined feedforward-feedback topology in accordance with one aspect of the present invention;
[0026] FIG.18 is a graph illustrating the active attenuation performance of each individual system/method in response to puretone noise; and
[0027] FIG. 19 is a graph illustrating experimentally determined maximum stable gains of the disclosed feedforward system and method with and without a feedback component. DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS
[0028] Operation of the adaptive feedforward LMS algorithm of the present invention is described in conjunction with the block diagram of Figure 1, which is an embodiment of an adaptive LMS filter 10 in the context of active noise reduction in a communication headset. In a feedforward noise reduction system, the external acoustic noise signal 12, Xk , is measured by a microphone 14. The external acoustic noise signal is naturally attenuated passively 16, as it passes through damping material, for example, a headset shell structure, and is absorbed by foam liners within the ear cup of the headset, as defined on [0061] .
[0029] The attenuated noise signal 18 is then cancelled by an equal and opposite acoustic noise cancellation signal 20, yk i generated using a speaker 22 inside the ear cup of the communication headset. The algorithm 24 that computes yk is the focus of the present invention. Termed an adaptive feedforward noise cancellation algorithm in the block diagram, it provides the cancellation signal as a function of the measured acoustic noise signal Xk (14'), and the error signal ek (26) , which is a measure of the residual noise after cancellation.
[0030] In real-world applications, each of these measured signals contains measurement noise due to microphones and associated electronics and digital quantization. Current embodiments of the adaptive feedforward noise canceling algorithm include two parameters - an adaptive step size μk that governs convergence of the estimated noise cancellation signal, and a leakage parameter λ . The traditional normalized, leaky feedforward LMS algorithm is given by the following two equations:
Figure imgf000009_0001
W = λWk + μ^kek
wherein Wk is a weight vector, or set of coefficients of a finite-impulse response filter. λ= 1 for ideal conditions: no measurement noise; no quantization noise; deterministic and statistically stationary acoustic inputs; discrete frequency components in Xk ; and infinite precision arithmetic. Under these ideal conditions, the filter coefficients converge to those required to minimize the mean-squared error ek . [0026] Algorithms for selecting parameter μk appear in the literature and modifications or embodiments of published μk selection algorithms appear in various prior art. However, the choice of parameters λ and μk as presented in the prior art does not guarantee stability of the traditional LMS algorithm under non-ideal real-world conditions, in which measurement noise in the microphone signals is present, finite precision effects reduce the accuracy of numerical computations, and noise fields are highly nonstationary. [0027] Furthermore, in current algorithms, the leakage parameter must be selected so as to maintain stability for worst case, i.e., nonstationary noise fields with impulsive noise content, resulting in significant noise cancellation degradation. Parameter λ is a constant between zero and one. Choosing λ = 1 results in aggressive performance, with compromised stability under real-world conditions. Choosing λ < 1 enhances stability at the expense of performance, as the algorithm operates far away from the optimal solution. [0028] The invention disclosed here is a computational method, based on a Lyapunov tuning approach, and its embodiment that automatically tunes time varying parameters λk and μk so as to maximize stability with minimal reduction in performance under noise conditions with persistent or periodic low signal-to-noise ratio, low excitation levels, and nonstationary noise fields. The automatic tuning method provides for time-varying tuning parameters λk and μk that are functions of the instantaneous measured acoustic noise signal Xk , weight vector length, and measurement noise variance.
[0029] The adaptive tuning law that arises from the Lyapunov tuning approach that has been tested experimentally is as follows:
Mk (3) (Xk + Qk)T(Xk + Qk)
(Xk + Qk)τ(Xk + Qk) -2Lσa 2 λ,r ( 4 : (Xk + Qk)T(Xk + Qk)
wherein Xk + Qk is the measured reference signal, which contains measurement noise Qk due to electronic noise and quantization. The measurement noise is of known variance σg 2 L is the length of weight vector Wk . This choice of tuning parameters provides maximal stability and performance of the leaky LMS algorithm, causing it to operate at small leakage factors only when necessary to preserve stability, while providing mean leakage factors near unity to maximize performance. Through application of these adaptive tuning parameters developed using the Lyapunov tuning approach, continual updating of the tuning parameters preserves stability and performance in non-ideal, real world noise fields described in [0005] .
Summary of Experimental Results [0030] Three candidate tuning laws that result from the Lyapunov tuning approach of the invention have been implemented and tested experimentally for low frequency noise cancellation in a prototype communication headset. The prototype headset consists of a shell from a commercial headset, which has been modified to include ANR hardware components, i.e., an internal error sensing microphone, a cancellation speaker, and an external reference noise sensing microphone. For experimental evaluation of the ANR prototype headset, the tuning method of the present invention is embodied as software within a commercial DSP system, the dSPACE DS 1103.
[0031] A block diagram 30, Fig. 2, shows one implementation of the present invention. The preferred embodiment of the 'Adaptive Leaky LMS' 24 contains a c- program that embodies the tuning method of the present invention, although a software implementation is not specific to nor a limitation of the present invention, but is applicable to all feedforward adaptive noise cancellation system embodiments. The three inputs to the Adaptive Leaky LMS block are the reference noise 14', the error microphone 26, and a reset' trigger 32 that is implemented for experimental analysis. The output signals are the acoustic noise cancellation signal 20, the tuned parameters λk (34) and μk (36), and the filter coefficients 38. [0032] The stability and performance of the resulting Active Noise Reduction (ANR) system has been investigated for a variety of noise sources ranging from deterministic discrete frequency components (pure tones) and stationary white noise to highly nonstationary measured F-16 aircraft noise over a 20dB dynamic range. Results demonstrate significant improvements in stability of the adaptive leaky LMS algorithm disclosed (Eq. 3-4) over traditional leaky or non-leaky normalized algorithms, while providing noise reduction performance equivalent to that of a traditional NLMS algorithm for idealized noise fields. Performance comparisons have been made as a function of signal-to-noise ratio (SNR) as well, showing a substantial improvement in ANR performance at low SNR. [0033] Performance of the prototype communication headset ANR system 40, Fig. 3, employing the disclosed tuning method has been experimentally compared with a commercial electronic noise cancellation headset that uses a traditional feedback ANR algorithm. Both headsets were evaluated within a low frequency test cell 42 specifically designed to provide a highly controlled and uniform acoustic environment. [0034] To perform the evaluation, a calibrated B&K microphone 44 was placed in the base of the test cell 42. A Larson-Davis calibrated microphone 46 with a wind boot was placed in the side 48 of the test cell 42, approximately 0.25 inches from the external reference noise microphone 50 of the headset 40 under evaluation. The Larson Davis microphone 46 measured the sound pressure level of the external noise when the headset 40 is in the test cell 42. The B&K microphone 44, which was mounted approximately at the location of a user's ear, was used to record sound pressure level (SPL) attenuation performance. With this test setup, each headset was subject to a sum of pure tones at 50, 63, 80, 100, 125, 160, and 200 Hz and 100 dB SPL. Both the passive attenuation and total attenuation were measured.
[0035] The active and passive attenuation of each headset, as measured by the power spectrum of the difference between the external Larson-Davis microphone 46 and internal B&K microphone 44 is recorded in Figure 4A and 4B respectively. The ANR prototype headset that uses the disclosed automatic tuning algorithm achieves superior active SPL attenuation at all frequencies in the 50-200 Hz band as measured at the B&K microphone 44. Passive noise attenuation of the commercial headset 52 is superior to the prototype headset 54, which being a prototype, was not optimized for passive performance. [0036] These measured results demonstrate that a headset with the combination of current technology in passive performance, and the superior active performance provided by the disclosed tuning method can achieve 30-35 dB SPL attenuation of low frequency stationary noise at the ear over the 50 to 200 Hz frequency band. This is a significant improvement over commercially available electronic feedback noise cancellation technology. There is both a theoretical and experimental basis for extending this performance over a wider frequency range. Additional test results are discussed below. Review of The Leaky Least Mean Square (LMS) Algorithm
[0037] A review of the LMS algorithm and its leaky variant follows. Denoting Xk eRn as the reference input at time t and dk ei? as the output of the unknown process, the LMS algorithm recursively selects a weight vector Wk ei? to minimize the squared error between dk and the adaptive filter output Wk T Xk .
The cost function is J = l (5) where ek = dk - W xk . (6)
The well-known Wiener solution, or optimum weight vector is
Figure imgf000015_0001
where E\XkX, is the autocorrelation of the input signal and
E^X dfc is the cross correlation between the input vector and process output. The Wiener solution reproduces the unknown process, such that dk = W0 Xk .
[0038] By following the stochastic gradient of the cost surface, the well-known unbiased, recursive LMS solution is obtained: Wk+ι = Wk +μekXk (8)
Stability, convergence, and random noise in the weight vector at convergence are governed by the step size μ. Fastest convergence to the Wiener solution is obtained for μ= ^max where m2X is the largest eigenvalue of the autocorrelation matrix .
Figure imgf000016_0001
[0039] As an adaptive noise cancellation method, LMS has some drawbacks. First, high input power leads to large weight updates and large excess mean-square error at convergence. Operating at the largest possible step size enhances convergence, but also causes large excess mean- square error, or noise in the weight vector, at convergence. A nonstationary input dictates a large adaptive step size for enhanced tracking, thus the LMS algorithm is not guaranteed to converge for nonstationary inputs.
[0040] In addition, real world applications necessitate the use of finite precision components, and under such conditions, the LMS algorithm does not always converge in the traditional form of eq. 4, even with an appropriate adaptive step size. Finally, nonpersistent excitation due to a constant or nearly constant reference input, such as can be the case during Λquiet periods' in adaptive noise cancellation systems with nonstationary inputs, can also cause weight drift.
[0041] In response to such issues, the leaky LMS (LLMS) algorithm or step-size normalized versions of the leaky LMS algorithm "leak off" excess energy associated with weight drift by including a constraint on output power in the cost function to be minimized. Minimizing the resulting cost function,
Figure imgf000017_0001
results in the recursive weight update equation Wk+1 = Wk +μekXk (10) where λ = l -γμ is the leakage factor. Under conditions of constant tuning parameters /land /, no measurement noise or finite-precision effects, and bounded signals J^ and e^ , eq. 6 converges to: k-\ . Wk = ∑ λ μXk_ι_iek_j.,. (11) ι=0 as &—»oo . Thus, for stability 0 < λ < 1 is required. The lower bound on λ assures that the sign of the weight vector does not change with each iteration.
[0042] The traditional constant leakage factor leaky LMS results in a biased weight vector that does not converge to the Wiener solution and hence results in reduced performance over the traditional LMS algorithm and its step size normalized variants.
[0043] The prior art documents a 60 dB decrease in performance for a simulated a leaky LMS over a standard LMS algorithm when operating under persistently exciting conditions. Hence, the need is to find time varying tuning parameters that maintain stability and retain maximum performance of the leaky LMS algorithm in the presence of quantifiable measurement noise and bounded dynamic range.
Lyapunov Tuning of the Leakage Factor [0044] In the presence of measurement noise Qk eR corrupting the reference signal^ , and with time varying leakage and step size parameters, λk andμk , the LLMS weight update equation becomes Wk+l = λkWk + μk(W0 TXk - ψξ(Xk + Qk))(Xk + Qk) (12 )
The stability analysis objective is to find operating bounds on the variable leakage parameter λk and the adaptive step size μk to maintain stability in the presence of noise vector Qk whose elements have known variance, given the dynamic range or a lower bound on the signal-to-noise ratio.
[0045] For stability at maximal performance, the present invention seeks time-varying parameters λ and μk such that certain stability conditions on a candidate Lyapunov function Vk are satisfied for all k in the presence of quantifiable noise on reference input Xk . Moreover, the choice of λk and μk should be dependent on measurable quantities, such that a parameter selection algorithm can be implemented in realtime. Finally, the selection algorithm should be computationally efficient. For uniform asymptotic stability, the Lyapunov stability conditions are: ϊ) Vk > 0 (13) ii) Vk+1 - Vk < 0 (14) and a decrescent Lyapunov function is required, i.e., Vk = 0 at Wk = 0 , and Vk < V forallAr≥O, where V is a time-invariant scalar function of Wk . Finally, for global uniform * asymptotic stability, the scalar function V must be radially unbounded, such that Hi) Jim Vk = ∞ (15) Wk→∞
Development of the candidate Lyapunov function proceeds by first defining^. = Wk ~ W0. Eq. 12 becomes Wk+1 = (λkI- μk(Xk + Qk)(Xk + Qk)T k + (λkI- I-μk(Xk + QύOty,, (16) Since scalar tuning parameters λ and μk are required, Wk and Wk+ι are projected in the direction of Xk+Qk, as shown in Figure 5:
Figure imgf000019_0001
wt+1- t+1lfc+all ll8) Combining Eq. 16 through 18 and simplifying the expression gives
**+l - itQfclfe +β/c]|j (19)
Figure imgf000019_0002
A candidate Lyapunov function satisfying stability condition i) above (Eq. 13) , is Vk = wk Twk (20) thus the Lyapunov function difference is Vk+1 - = k+ k+l - wkwk (21)
The expression for the projected weight update in Eq. 19 can be simplified as wk+l = ^kWk + rιkW0)uk+r2lw k (22) where
Mt"fc÷al (23) is the unit vector in the direction of Xk+Qk, and k = kk(Xk+Qkf(Xk + Qk) (24) Yχk = λk-l (25) nk = -μkXk + Qk)τ(xk + Qk) (26) ak = » ^ .I (27)
[0046] With these definitions, the Lyapunov function difference becomes,
VM~Vk = (l -ϊ)Wk uku{ Wk+γ W:uku{W02'Waka{W0 (28) + 2φkχh Wk uku .{TTWTΓ0 + 2φkγ2k Wk1
Figure imgf000020_0001
Note that the projected weight vector of Eq. 17 and 18 and the resulting Lyapunov function candidate of Eq. 20 do not satisfy condition Lyapunov stability condition Hi) (Eq. 15) , which is required for global uniform asymptotic stability.
However, it is possible to find a time-invariant scalar * * function V such that the Lyapunov candidate Vk < V for all k >0. [0047] Since the scalar projection is always in the direction of the unit vector defined by eq. 16, an example of such a function is V =l0W Wk. Hence, the Lyapunov function can be used to assess uniform asymptotic stability. [0048] Note also that there are two conditions that may be considered problematic with the projected weight vector.
These occur if (a) Xk = -Qk or (b) Wk is orthogonal to u or some component of Wk is orthogonal to uk. Condition (a) is highly unlikely, especially at realistic tap lengths and signal-to-noise ratios (SNR) . In fact, if this condition does occur, then, intuitively, it must be the case that SNR is so low that noise cancellation is futile, since the noise floor effectively dictates the maximum performance that can be achieved. [0049] If W is orthogonal to uk under reasonable SNR conditions, then it is likely that the filter output ek is very close to zero, i.e., the LMS algorithm is simply unnecessary if such a condition persists. Thus, though it is possible, but unlikely, that one or more of the weight vector components could become unbounded, in considering such unlikely occurrences it is impossible to avoid serious performance degradation.
[0050] The goal of the Lyapunov analysis is to enable quantitative comparison of stability and performance tradeoffs for candidate tuning rules. Since uniform asymptotic stability suffices to make such comparisons, and since the Lyapunov function of Eq. 20 enhances the ability to make such comparisons, it was selected for the analysis that follows . [0051] Several approaches to examining Lyapunov stability condition ii) Vk+\ - Vk < 0 for Eq. 28 exist. The usual approach to determining stability is to examine Vk . - Vk term by term to determine whether the two parameters λk vxA μk can be chosen to make each term negative thereby guaranteeing uniform asymptotic stability. Since there are several terms that are clearly positive in Eq. 28, there is no guarantee that each individual term will be negative. Furthermore, it is clear from an analysis of Eq. 28 that the solution is nearly always biased away from zero. At Wk = W - Wo = 0 , Eq. 28 becomes: +l ~
Figure imgf000022_0001
+ 2γlkγ2k W0 Tukak TW0 (29) For 0 < λ < 1, all coefficients of terms in Eq. 29 are positive, and it is clear that a negative definite Vk+\ — Vk results only if Yιk W ukuk W0 + γ kW akak W0 < -2γlkγlk W/uk k W0 with YlkY2j > 0- That the leaky LMS algorithm, as examined using the Lyapunov candidate of Eq. 20, is biased away from W0 is in agreement with the prior art. It is possible, but difficult, to examine the remaining space of Wk = Wk — W0 (i.e., the space that excludes the origin) to determine whether time varying tuning parameters can be found to guarantee stability of some or all other points in the space or a maximal region of the space. [0052] Time varying tuning parameters are required since constant tuning parameters found in such a manner will retain stability of points in the space at the expense of performance. However, since we seek time varying leakage and step size parameters that are uniquely related to measurable quantities and since the Wiener solution is generally not known a priori, the value of such a direct analysis of the remaining space of Wk = Wk - W0 is limited.
[0053] Thus, the approach taken in the present invention is to define the region of stability around the Wiener solution in terms of parameters:
A - k (30) W0 uk , Wlak B = °τ k (31) W0 uk and to parameterize the resulting Lyapunov function difference such that the remaining scalar parameter (s) can be chosen by optimization. [0054] The parameters A and B physically represent the output error ratio between the actual output and ideal output for a system converged to the Wiener solution, and the output noise ratio, or portion of the ideal output that is due to noise vector Qk . Physically, these parameters are inherently statistically bounded based on i) the maximum output that a real system is capable of producing, ii) signal-to-noise ratio in the system, and iii) the convergence behavior of the system. Such bounds can be approximated using computer simulation. These parameters provide convenient means for visualizing the region of stability around the Wiener solution and thus for comparing candidate tuning rules. [0055] In a persistently excited system with high signal- to-noise ratio, B approaches zero, while the Wiener solution corresponds to A = 0, i.e., Wk = W0 . Thus, high performance and high SNR operating conditions imply both A and B are near zero in the leaky LMS algorithm, though the leaky solution will always be biased away from A = 0. In a system with low excitation and/or low signal-to-noise ratio, larger instantaneous magnitudes of A and B are possible, but it is improbable that the magnitude of either A or B is »1 in practice. Note that B depends only on the reference and noise vectors, and thus it cannot be influenced by the choice of tuning parameters. B can, however, affect system stability.
[0056] Using parameters A and B, Eq. 28 becomes
Vk+ι - vk = ( 32 )
Figure imgf000024_0001
By choosing an adaptive step size and/or leakage parameter that simplifies analysis of Eq. 32, one can parameterize and subsequently determine conditions on remaining scalar parameters such that Vk — Vk < 0 for the largest region possible around the Wiener solution. Such a region is now defined by parameters A and B, providing a means to graphically display the stable region and to visualize performance/stability tradeoffs introduced for candidate leakage and step size parameters.
Comparison of Candidate Tuning Laws using Lyapunov Analysis [0057] To demonstrate the use of the parameterized Lyapunov difference of Eq. 32, consider three candidate leakage parameter and adaptive step size combinations. [0058] The first candidate uses a traditional choice for leakage parameter in combination with a traditional choice for adaptive step size to provide:
Figure imgf000024_0002
2 wherein σq is the variance of quantifiable noise corrupting each component of vector Xk . This choice results in a simple relationship for the constants in Eq. 32
Figure imgf000025_0001
Y2k = ' o (36)
Thus, the combined candidate step size and leakage factor parameterize Eq. 32 in terms of μ0 . [0059] To determine the optimal 0, one can perform a scalar optimization of Vk+ι — Vk with respect to μ0 and evaluate the result for worst-case constants A and J3. In essence, one seeks the value of μ0 that makes Vk .\ - Vk most negative for worst-case deviations of weight vector Wk from the Wiener solution and for worst-case effects of measurement noise Qk . Worst case A and B are chosen to be that combination in the range Am±n < A < 0 and 0 < A < Amax, Bm±n < B < Bmax that provides the smallest (i.e., most conservative) step size parameter μ0 . [0060] For example, for Amin = Bmin = -1 and Amax = Bmax = 1, and the traditional adaptive leakage parameter and step size combination of Eq. 33 and 34, this optimization procedure results in μ0 = 1/3, which is consistent with the choice for μ0 . [0061] The second candidate also retains the traditional leakage factor of Eq. 34, and finds an expression for μk as a function of the measured reference input and noise covariance directly by performing a scalar optimization of Vk+ι ~ Vk with respect to μk . Again, the results are evaluated for worst- case conditions on A and B, as described above. This scalar optimization results in
Figure imgf000026_0001
[0062] The final candidate appeals to the structure of Eq. 32 to determine an alternate parameterization as a function of μ0. Selecting
Figure imgf000026_0002
results in φk = (l-μ0k (40)
Figure imgf000026_0003
The expression for λk in Eq. 39 is not measurable, but it can be approximated as
Figure imgf000026_0004
wherein L is the filter length.
[0063] Equation 43 is a function of statistical and measurable quantities, and is a good approximation of Eq. 39 when
Figure imgf000026_0005
■ The corresponding definitions of Φ Y\k Ylk> ∞&Mkr Ec3- 32 becomes
Figure imgf000026_0006
The optimum μ0 for this candidate, which is again found by scalar optimization subject to worst case conditions on A and β is μ0 = 1/2.
[0064] In summary, the three candidate adaptive leakage factor and step size solutions are Candidate 1: Eq. 33 and 34, Candidate 2: Eq. 33 and 37, and Candidate 3: Eq. 38 and 43. All are computationally efficient, requiring little additional computation over a fixed leakage, normalized LMS algorithm, and all three candidate tuning laws can be implemented based on knowledge of the measured, noise corrupted reference input, the variance of the measurement noise, and the filter length.
[0065] To evaluate stability and performance tradeoffs, one examines Vk+ι ~ Vk for various instantaneous signal-to-noise ratios |ΛrJ/|βJ (SNR) , and 1 > A > -1, 1 > B > -1 .
[0066] Figure 6 shows plots of V\ - Vk vs. A and B for SNR of 2, (Figs. 6A-6C) 10 (Figs. 6D-6F) , and 100 (Figs. 6G-6I) , and a filter length of 20. Numerical results corresponding to Figure 6 are shown in Figure 7. Figure 6 includes the zero' plane, such that stability regions provided by the intersection of the Lyapunov difference with this plane can be visualized.
[0067] Note again, that A = 0 corresponds to the LMS Wiener solution. At sufficiently high SNR, for all candidates, Vk+ι - Vk = 0 for A = B = 0, i.e., operation at the Wiener solution with Q =0 . A notable exception to this is candidate 3, for which Vk+\ — Vk > 0 for A = 0 and B = 0 and SNR = 2, due to the breakdown of the approximation of the leakage factor in Eq. 43 for low SNR.
[0068] For A = 0 and B > 0, the Wiener solution is unstable, which is consistent with the bias of leaky LMS algorithms away from the Wiener solution. The uniform asymptotic stability region in Figure 6 is the region for which Vk+\ - Vk < 0. At sufficiently high SNR, this stability region is largest for candidate 3, followed by candidate 1. Candidate 2 provides the smallest overall stability region. [0069] For example, if one takes a slice of each Figure 6 at B = -1, the resulting range of A for which Vk+\ - Vk > 0 is largest for candidate 2. However, the likelihood of obtaining such combinations of A and B in practice is remote for sufficiently high SNR and a stationary or slowly time varying Wiener solution. Near the origin, which is the most likely operating point, the stability region for all three candidates is similar for sufficiently high SNR. [0070] Performance of each candidate tuning law is assessed by examining both the size of the stability region and the gradient of Vk+γ - Vk with respect to parameters A and B. Note from Eq. 32 that the gradient of Vk+- - Vk approaches zero as λk approaches one and//^ approaches zero (i.e., stability, but no convergence) . In the stable region of Figure 6, the gradient of the Lyapunov difference is larger for tuning that provides an aggressive step size.
[0071] Thus, a tuning law providing a more negative V.\ - Vk in the stable region should provide the best performance, while the tuning law providing the largest region in which Vk+\ - V < 0 provides the best stability. Figure 7 records the maximum and minimum values of Vk+\ - Vk for the range of A and B examined, showing candidate 2 should provide the best performance (and least stability) , while candidate 3 provides the best overall stability/performance tradeoff for high SNR, followed by candidates 1 and 2.
[0072] For all three candidates, leakage factor approaches one as signal-to-noise ratio increases, as expected, and candidate 2 provides the most aggressive step size, which relates to the larger gradient of Vk+γ - Vk and thus the best predicted performance. An alternate view of Vk+ι - Vk as it relates to performance is to consider Vk+ — Vk as the rate of change of energy of the system. The faster the energy decreases, the faster convergence, and hence the better performance.
[0073] The results of this stability analysis do not require a stationary Wiener solution, and thus these results can be applied to reduction of both stationary and nonstationary Xk . The actual value of the Wiener solution, which is embedded in the parameters A and B does affect the stability region, and it is possible, that any of the three candidates can be instantaneously unstable given an inappropriate combination of A and B. [0074] Nevertheless, it is appropriate to use the graphical representation of Figure 6 to determine how close to the Wiener solution one can operate as a measure of performance and to use the size of the stability region as a measure of stability. In cases where the Wiener solution is significantly time variant, the possibility of operating far from the Wiener solution increases, requiring more attention to developing candidate tuning laws that enhance the stability region for larger magnitudes of parameters A and B. Experimental results [0075] The three candidate Lyapunov tuned leaky LMS algorithm are evaluated and compared to i) an empirically tuned, fixed leakage parameter leaky, normalized LMS algorithms (LNLMS) , and ii) an empirically tuned normalized LMS algorithm with no leakage parameter (NLMS) . The comparisons are made for a low-frequency single-source, single-point noise cancellation system in an acoustic test chamber (42, Fig. 3) designed to provide a highly controlled and repeatable acoustic environment with a flat frequency response over the range of 0 to 200 Hz for sound pressure levels up to 140 dB.
[0076] The system under study is a prototype communication headset earcup. The earcup contains an external microphone to measure the reference signal, an internal microphone to measure the error signal, and an internal noise cancellation speaker to generate yk . Details regarding the prototype are given above in connection with Figure 3. [0077] The reference noise is from an F-16, a representative high-performance aircraft that exhibits highly nonstationary characteristics and substantial impulsive noise content. The noise source is band limited at 50 Hz to maintain a low level of low frequency distortion in the headset speaker and 200 Hz, the upper limit for a uniform sound field in the low frequency test cell. [0078] Figure 8 shows the low frequency regime of the reference noise power spectrum along with statistically determined upper and lower bounds on the power spectrum that indicate the degree of nonstationarity of the noise source. To obtain these bounds, the variation in the power spectral density (PSD) of a three-second-noise sample was calculated. The three-second sample was then divided into 100 equal length segments, and the PSD of each 0.03-second segment was determined. From these sampled spectrums, the minimum and maximum PSD as a function of frequency was determined, providing upper and lower bounds on the power spectrum. [0079] The noise floor of the test chamber 42 is 50 dB . Without active noise cancellation, the earmuff provides approximately 5 dB of passive noise reduction over the 50 to 200 Hz frequency band. The amplitude of the reference noise source is established to evaluate algorithm performance over a 20 dB dynamic range, i.e., sound pressure levels of 80 dB and 100 dB, as measured inside the earcup after passive attenuation. The difference in sound pressure levels tests the ability of the tuned leaky LMS algorithms to adapt to different signal-to-noise ratios.
[0080] The two noise amplitudes represent signal-to-noise ratio (SNR) conditions for the reference microphone measurements of 35 dB and 55 dB, respectively. For the F-16 noise source and 100 dB SPL (55 dB SNR), analysis of V.γ - Vk of Eq. 32 for Lyapunov tuned candidates shows statistically determined bounds on B of -0.6 < B < 0.6, while for the 80 dB SPL (35 dB SNR) , statistically determined bounds on B are -3 < B < 3. Thus, Figure 6, which gives the Vk .γ - Vk surface for each candidate algorithm, shows that by lowering SNR to 35 dB, instability is possible for all three candidates, as the fixed step size is chosen for worst case conditions on B of - 1 < B < 1. [0081] Thus, in addition to eliciting stability and performance tradeoffs, the 80 dB SPL noise source tests the limits of stability for the three candidate algorithms. The quantization noise magnitude is 610e-6 V, based on a 16-bit round-off A/D converter with a +10 V range and one sign bit. The candidate LMS algorithms are implemented experimentally using a dSPACE DS1103 DSP board. A filter length of 250 and weight update frequency of 5 kHz are used. The starting point for the noise segments used in the experiments is nearly identical for each test, so that noise samples between different tests overlap.
[0082] In the first part of this comparative study, the empirically tuned NLMS and LNLMS filters with constant leakage parameter and the traditional adaptive step size of Eq. 34 are tuned for the 100 dB SPL and subsequently applied without change to the system for the 80 dB SPL. On the other hand, the constant leakage parameter LNLMS filter is empirically tuned for 80 dB and subsequently applied to the 100 dB SPL test condition. [0083] These two empirically tuned algorithms are denoted LNLMS (100) and LNLMS (80), respectively. For both filters, μ0 - 1/3, and the respective leakage parameter is given in Figure 9. Application of the algorithm tuned for a specific SPL to cancellation of noise not matching the tuning conditions demonstrates the loss of performance that results under constant tuning parameters that would be required for a noise cancellation system subject to this 20 dB dynamic range. In all experiments, the weight vector elements are initialized as zero. [0084] Figure 10 shows experimental results for these three filters (NLMS, LNLMS (100), and LNLMS (80)) operating at 100 dB SPL. Of the empirically tuned filters, the NLMS algorithm and the LNLMS tuned for 100 dB algorithm show similar performance, while the LNLMS algorithm tuned for 80 dB shows significant performance reduction at steady-state. Here, SNR is sufficiently high that only a small amount of leakage is required to guarantee stability, thus performance degradation due to the leakage factor is minimal. Note that although the NLMS algorithm is stable after five seconds of operation, a slow weight drift occurs, such that the leakage factor is required.
[0085] Figure 11 shows results for the 80 dB SPL. Here, the low SNR causes weight instability in the NLMS algorithm during the five second experiment. The mismatch in tuning conditions, i.e., using the LNLMS (100) algorithm under 80 dB SPL conditions also results in weight drift instability. Evidence of instability of the NLMS and LNLMS (100) algorithms at 80 dB is shown in time histories of the root-mean square (RMS) weight vector in Figures 12A and 12B. The results of Figures 10 through 12 demonstrate both the loss of stability when using an overly aggressive (large) fixed parameter leakage parameter and the loss of performance when a less aggressive (small) leakage parameter is required in order to retain stability over large changes in the dynamic range of the reference input signal.
[0086] The Lyapunov based tuning approach provides a candidate algorithm that retains stability and satisfactory performance in the presence of the nonstationary noise source over the 20 dB dynamic range, i.e., at both 80 and 100 dB SPL. Figure 13 shows performance at 100 dB SPL, and Figure 14 shows performance at 80 dB SPL. [0087] At 100 dB SPL (Figure 13), all three candidate algorithms retain stability, and at steady-state, noise reduction performance of all three candidate algorithms exceeds that of empirically tuned leaky LMS algorithms. In fact, performance closely approximates that of the NLMS algorithm, which represents the best possible performance for a stable system, as it includes no performance degradation due to a leakage bias.
[0088] At 80 dB SPL (Figure 14), candidates 2 and 3 are unstable at 80 dB SPL, reflecting the fact that candidate algorithms do not necessarily guarantee uniform asymptotic stability when assumptions regarding bounds on measurement noise are exceeded. Candidate 3, which was predicted by Lyapunov analysis to provide the best stability characteristics of the three candidates retains stability and provides a steady-state SPL attenuation exceeding that of the LNLMS (80) by 5 dB.
[0089] Since the LNLMS (80) is the best performing stable fixed leakage parameter algorithm available, the performance improvement is significant. Note that comparison of performance at 80 dB SPL to the NLMS algorithm cannot be made, because the NLMS algorithm is unstable for the 80 dB SPL (35 dB SNR) .
[0090] Figure 15 shows the RMS weight vector histories for both 80 dB and 100 dB reference input sound pressure levels, providing experimental evidence of stability of all three candidates at 100 dB SPL and of candidate 3 at 80 dB SPL. [0091] Performance gains of Lyapunov tuned candidates over the fixed leakage parameter LMS algorithms are confirmed by the mean and variance of the leakage factor for each candidate, as shown in Fig. 9. For all three candidates, the variance of the leakage factor is larger for the 80 dB test condition that for the 100 dB condition, as expected, since the measured reference signal at 80 dB represents lower average and instantaneous signal-to-noise ratios. Moreover, with the exception of candidate 1 at 80 dB, the mean leakage factor is larger than that provided by empirical tuning. [0092] Hence, on average, the Lyapunov tuned LMS algorithms are more aggressively tuned and operate closer to the Wiener solution, providing better performance over a large dynamic range than constant leakage factor algorithms. [0093] Finally, relative performance, which is predicted to be most aggressive for candidate 2, followed by candidates 3 and 1, respectively, is seen in Figure 14. Candidate 2 provides the fastest convergence and the largest SPL attenuation of the three candidates.
[0094] The experimental results provide evidence that the method of tuning an adaptive Leaky LMS Filter according to the algorithm of the present invention provides stability and performance gains which result in the reduction of highly nonstationary noise for an optimized combination of both adaptive step size and adaptive leakage factor without requiring empirical tuning, with candidate 3 providing the best overall stability and performance tradeoffs. [0095] In accordance with another aspect of the present invention, hybridization of a traditional feedback control law with a feedforward control law improves ANR performance and stability margins. The Lyapunov-tuned feedforward controller described herein has excellent response in systems with time-varying signal-to-noise ratio. Acting alone, the algorithm (s) disclosed above substantially improves ANR performance over traditional LMS filters and exhibits excellent performance for non-stationary noise sources, and good performance for non-stationary noise sources. [0096] Figure 17 shows a hybrid feedforward-feedback ANR topology in accordance with the present invention. A reference microphone 100 measures the primary source, which enters the unknown acoustic process H(z) 102, and the error signal 104 remaining after ANR is measured by a microphone 106. In the feedforward component, an adaptive LMS filter, provides a cancellations signal -y , 108. Here, the feedforward system can be thought of as providing a smaller error signal for the feedback controller to act on, since it models the unknown acoustic process, and thus the system can tolerate an overall increase in the feedback or feedforward controller gain without destabilizing the system. Alternatively, one can consider a feedback controlled system as being acted upon by the feedforward controller, which because it is adaptive, performs its task whether or not the feedback control component is in place.
[0097] In experimental evaluation of the architecture, a broadband, feedback controller providing 5-10 dB of attenuation in the 40 Hz to 1600 Hz frequency band is paired with the feedforward controller, which is tuned according to one aspect of the present invention. Both the feedback and feedforward components are implemented digitally. Because of this, no additional hardware components are required to add the feedback component beyond those used for the feedforward controller. Figure 18 shows sample experimental results. At low frequencies (<100 Hz) , the feedback component provides 1- 8 dB of active attenuation, and the feedforward component, which is tuned according to method disclosed herein provides 15-27 dB of attenuation. However, the hybrid system demonstrates overall performance that is greater than the sum of the individual components at frequencies below 80 Hz. The exceptional performance of the hybrid system is achieved by pairing the feedforward controller tuned in accordance with the method disclosed herein with the traditional infinite impulse response feedback controller.
[0098] It is known that feedback controllers exhibit less sensitivity than feedforward controllers to noise source characteristics. Thus, for non-stationary noise sources, the hybrid system exhibits the positive characteristics of the
Lyapunov-tuned feedforward system combined with the positive characteristics of a feedback controller in exhibiting less sensitivity to noise source characteristics. [0099] Figure 19 shows measured stability margins of a hybrid controller from experimental evaluation of the system when applied to ANR in a hearing protector. Measurements were made using the low frequency acoustic test cell and digital signal processing development system described herein. Stability margin is measured by the tolerable increase in the feedforward controller gain ( Kff ) before the system shows evidence of instability with and without the feedback component in place. With the hybrid system, gain margin improves by a factor of 2 to over 1000 through the band evaluated.
[00100] It is important to note that the present invention is not intended to be limited to a device or method which must satisfy one or more of any stated or implied objects or features of the invention. It is also important to note that the present invention is not limited to the preferred, exemplary, or primary embodiment (s) described herein. Modifications and substitutions by one of ordinary skill in the art are considered to be within the scope of the present invention, which is not to be limited except by the following claims. [00101] The invention claimed is:

Claims

CLAIMS 1. A method of tuning an adaptive feedforward noise cancellation algorithm, comprising the acts of: providing a feedback active noise reduction (ANR) circuit, for providing an ANR error signal; providing a feedforward LMS tuning algorithm including at least first and second time varying parameters wherein said feedforward LMS tuning algorithm includes the formulas: yk = wξxk Wk+ l = λkWk + μkXkek ' an adjusting said at least first and second time varying parameters as a function of instantaneous measured acoustic noise, a weight vector length and measurement noise variance, wherein said time varying parameters include: μk = frk + QkYfrk +Qk) λ Xk +Qk)T(Xk +Qk)-2Lσ2 (x* +Q*F(x* +Q*)
wherein Xk = X]ir + Q]!: is a measured reference signal; Q^ is measurement noise, including electronic noise and quantization noise;
2 σq is the known or measured variance of the measurement noise; L is the length of the LMS weight vector Wk ; and e^is an error signal which is the net result of both the feedforward method and the feedback circuit.
2. The method of claim 1 wherein the output of the filter yk is multiplied by a feedforward proportionality constant Kft to produce a feedforward acoustic noise cancellation signal Kβ-yk and wherein the error signal ek is acted on by a digital infinite impulse response filter so as to produce a cancellation signal rk , which is multiplied by a feedback proportionality constant Kβ, and the sum of the feedforward and feedback component s Kjryk + Kβrk provides a total noise cancellation signal .
3. The method of claim 2 wherein said ANR error signal of feedback active noise reduction (ANR) circuit and said feedforward LMS tuning algorithm each provide an active noise reduction performance value that is greater than a sum of said ANR circuit and said LMS tuning algorithm.
- 31
4. A method of tuning an algorithm for providing noise cancellation, comprising the acts of: receiving a measured reference signal, the measured reference signal including a measurement noise component having a measurement noise value of known variance; and generating an acoustic noise cancellation signal according to the formulas: yk = wk 1xk Wk+\ = λkWk + μkXkek
wherein time varying parameters λk and μk are determined according to the formulas : βoλk k {Xk+Qk)T{Xk+Qk) λ _(Xk+Qk)T(Xk+Qk)-2Lσ2 (Xt+QtFfø+Qjfc) wherein Xk = X +Qk ^s a measured reference signal;
Q^ is electronic noise and quantization; σq 2 is a known variance of the measurement noise;
L is the length of weight vector Wk; and ek is an error signal which is the net result of both a feedforward tuning method and a feedback active noise reduction method.
5. A method of tuning a least mean square (LMS) filter comprising the acts of: providing a feedback active noise reduction (ANR) circuit, for providing an ANR error signal; formulating a Lyapunov function of a LMS filter weight vector, a reference input signal, a measurement noise on the measured reference input signal, a time varying leakage parameter λk , and a step size parameter μk ; using the resultant Lyapunov function to identify formulas for computing the time varying leakage parameter λk and step size parameter μk that maximize stability and performance of the resultant LMS filter weight vector update equation Wk+ι = λkWk + μkekXk
wherein said time varying parameters determined are
Figure imgf000042_0001
Figure imgf000042_0002
wherein Xk = X + Qk is a measured reference signal; Q^ is electronic noise and quantization; σ 2 is a known variance of the measurement noise; L is the length of weight vector Wk ; and ek is an error signal which is the net result of both the ANR circuit and the LMS filter.
PCT/US2005/012598 2004-05-10 2005-04-13 Tuned feedforward lms filter with feedback control WO2005112849A2 (en)

Priority Applications (2)

Application Number Priority Date Filing Date Title
EP05758737A EP1744713A4 (en) 2004-05-10 2005-04-13 Tuned feedforward lms filter with feedback control
JP2007513154A JP2007536877A (en) 2004-05-10 2005-04-13 Tuned feedforward LMS filter with feedback control

Applications Claiming Priority (2)

Application Number Priority Date Filing Date Title
US10/842,714 2004-05-10
US10/842,714 US6996241B2 (en) 2001-06-22 2004-05-10 Tuned feedforward LMS filter with feedback control

Publications (2)

Publication Number Publication Date
WO2005112849A2 true WO2005112849A2 (en) 2005-12-01
WO2005112849A3 WO2005112849A3 (en) 2006-01-12

Family

ID=35428819

Family Applications (1)

Application Number Title Priority Date Filing Date
PCT/US2005/012598 WO2005112849A2 (en) 2004-05-10 2005-04-13 Tuned feedforward lms filter with feedback control

Country Status (5)

Country Link
US (1) US6996241B2 (en)
EP (1) EP1744713A4 (en)
JP (1) JP2007536877A (en)
KR (1) KR20070010166A (en)
WO (1) WO2005112849A2 (en)

Cited By (18)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
JP2007241104A (en) * 2006-03-10 2007-09-20 Saitama Univ Adaptive linear prediction unit, speech enhancing device, and speech enhancing system
JP2008116782A (en) * 2006-11-07 2008-05-22 Sony Corp Noise canceling system and noise canceling method
GB2445388A (en) * 2007-02-16 2008-07-09 Sonaptic Ltd Noise-canceling ear-worn speaker devices with vents to bypass seals
JP2008250270A (en) * 2007-03-02 2008-10-16 Sony Corp Signal processing apparatus and signal processing method
WO2010129241A1 (en) * 2009-04-28 2010-11-11 Bose Corporation Dynamically configurable anr filter and signal processing topology
US8073151B2 (en) 2009-04-28 2011-12-06 Bose Corporation Dynamically configurable ANR filter block topology
US8073150B2 (en) 2009-04-28 2011-12-06 Bose Corporation Dynamically configurable ANR signal processing topology
US8090114B2 (en) 2009-04-28 2012-01-03 Bose Corporation Convertible filter
US8165313B2 (en) 2009-04-28 2012-04-24 Bose Corporation ANR settings triple-buffering
US8184822B2 (en) 2009-04-28 2012-05-22 Bose Corporation ANR signal processing topology
JP5114611B2 (en) * 2007-09-28 2013-01-09 株式会社DiMAGIC Corporation Noise control system
JP2013037381A (en) * 2012-10-17 2013-02-21 Sony Corp Noise canceling device and noise canceling method
EP2597889A1 (en) 2011-11-22 2013-05-29 Parrot Headphones with non-adaptive active noise control
US8693699B2 (en) 2008-07-29 2014-04-08 Dolby Laboratories Licensing Corporation Method for adaptive control and equalization of electroacoustic channels
CN105785349A (en) * 2016-05-09 2016-07-20 浙江大学 Noise removing method for phased-array three dimensional acoustic photographing sonar
CN109697986A (en) * 2018-09-19 2019-04-30 四川大学 A kind of adaptive deviation compensation echo cancel method based on minimum cube absolute value
CN110610693A (en) * 2019-08-09 2019-12-24 漳州立达信光电子科技有限公司 Weighted mixed type active anti-noise system and controller
WO2020241915A1 (en) * 2019-05-29 2020-12-03 이대훈 Bone conduction bluetooth mono earset

Families Citing this family (107)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US8019091B2 (en) 2000-07-19 2011-09-13 Aliphcom, Inc. Voice activity detector (VAD) -based multiple-microphone acoustic noise suppression
US8280072B2 (en) 2003-03-27 2012-10-02 Aliphcom, Inc. Microphone array with rear venting
US9066186B2 (en) 2003-01-30 2015-06-23 Aliphcom Light-based detection for acoustic applications
US9099094B2 (en) 2003-03-27 2015-08-04 Aliphcom Microphone array with rear venting
US7308106B2 (en) * 2004-05-17 2007-12-11 Adaptive Technologies, Inc. System and method for optimized active controller design in an ANR system
US20070041606A1 (en) * 2005-08-22 2007-02-22 David Clark Company Incorporated Apparatus and method for noise cancellation in communication headset using dual-coil speaker
US8059828B2 (en) * 2005-12-14 2011-11-15 Tp Lab Inc. Audio privacy method and system
US20070154049A1 (en) * 2006-01-05 2007-07-05 Igor Levitsky Transducer, headphone and method for reducing noise
GB2446966B (en) 2006-04-12 2010-07-07 Wolfson Microelectronics Plc Digital circuit arrangements for ambient noise-reduction
US8275120B2 (en) * 2006-05-30 2012-09-25 Microsoft Corp. Adaptive acoustic echo cancellation
WO2009042635A1 (en) * 2007-09-24 2009-04-02 Sound Innovations Inc. In-ear digital electronic noise cancelling and communication device
CN101400007A (en) * 2007-09-28 2009-04-01 富准精密工业(深圳)有限公司 Active noise eliminating earphone and noise eliminating method thereof
JP4506873B2 (en) * 2008-05-08 2010-07-21 ソニー株式会社 Signal processing apparatus and signal processing method
US8699721B2 (en) * 2008-06-13 2014-04-15 Aliphcom Calibrating a dual omnidirectional microphone array (DOMA)
JP5228647B2 (en) * 2008-06-19 2013-07-03 ソニー株式会社 Noise canceling system, noise canceling signal forming method, and noise canceling signal forming program
US8699719B2 (en) * 2009-03-30 2014-04-15 Bose Corporation Personal acoustic device position determination
US8532310B2 (en) 2010-03-30 2013-09-10 Bose Corporation Frequency-dependent ANR reference sound compression
WO2010129272A1 (en) * 2009-04-28 2010-11-11 Bose Corporation Sound-dependent anr signal processing adjustment
US8345888B2 (en) * 2009-04-28 2013-01-01 Bose Corporation Digital high frequency phase compensation
US8472637B2 (en) 2010-03-30 2013-06-25 Bose Corporation Variable ANR transform compression
US8085946B2 (en) * 2009-04-28 2011-12-27 Bose Corporation ANR analysis side-chain data support
US8611553B2 (en) 2010-03-30 2013-12-17 Bose Corporation ANR instability detection
JP5293817B2 (en) * 2009-06-19 2013-09-18 富士通株式会社 Audio signal processing apparatus and audio signal processing method
US8385559B2 (en) * 2009-12-30 2013-02-26 Robert Bosch Gmbh Adaptive digital noise canceller
KR100987981B1 (en) * 2010-03-26 2010-10-18 삼성탈레스 주식회사 Apparatus and method for distinguishing between activity signal and transition noise
JP2012023637A (en) * 2010-07-15 2012-02-02 Audio Technica Corp Noise cancel headphone
US8908877B2 (en) 2010-12-03 2014-12-09 Cirrus Logic, Inc. Ear-coupling detection and adjustment of adaptive response in noise-canceling in personal audio devices
CN103270552B (en) 2010-12-03 2016-06-22 美国思睿逻辑有限公司 The Supervised Control of the adaptability noise killer in individual's voice device
US8718291B2 (en) * 2011-01-05 2014-05-06 Cambridge Silicon Radio Limited ANC for BT headphones
DE102011013343B4 (en) * 2011-03-08 2012-12-13 Austriamicrosystems Ag Active Noise Control System and Active Noise Reduction System
US9214150B2 (en) 2011-06-03 2015-12-15 Cirrus Logic, Inc. Continuous adaptation of secondary path adaptive response in noise-canceling personal audio devices
US9076431B2 (en) 2011-06-03 2015-07-07 Cirrus Logic, Inc. Filter architecture for an adaptive noise canceler in a personal audio device
US8848936B2 (en) 2011-06-03 2014-09-30 Cirrus Logic, Inc. Speaker damage prevention in adaptive noise-canceling personal audio devices
US8948407B2 (en) 2011-06-03 2015-02-03 Cirrus Logic, Inc. Bandlimiting anti-noise in personal audio devices having adaptive noise cancellation (ANC)
US9318094B2 (en) 2011-06-03 2016-04-19 Cirrus Logic, Inc. Adaptive noise canceling architecture for a personal audio device
US8958571B2 (en) 2011-06-03 2015-02-17 Cirrus Logic, Inc. MIC covering detection in personal audio devices
US9824677B2 (en) 2011-06-03 2017-11-21 Cirrus Logic, Inc. Bandlimiting anti-noise in personal audio devices having adaptive noise cancellation (ANC)
CN102348151B (en) * 2011-09-10 2015-07-29 歌尔声学股份有限公司 Noise canceling system and method, intelligent control method and device, communication equipment
US9325821B1 (en) * 2011-09-30 2016-04-26 Cirrus Logic, Inc. Sidetone management in an adaptive noise canceling (ANC) system including secondary path modeling
US9142205B2 (en) 2012-04-26 2015-09-22 Cirrus Logic, Inc. Leakage-modeling adaptive noise canceling for earspeakers
US9014387B2 (en) 2012-04-26 2015-04-21 Cirrus Logic, Inc. Coordinated control of adaptive noise cancellation (ANC) among earspeaker channels
US9082387B2 (en) 2012-05-10 2015-07-14 Cirrus Logic, Inc. Noise burst adaptation of secondary path adaptive response in noise-canceling personal audio devices
US9318090B2 (en) 2012-05-10 2016-04-19 Cirrus Logic, Inc. Downlink tone detection and adaptation of a secondary path response model in an adaptive noise canceling system
US9319781B2 (en) 2012-05-10 2016-04-19 Cirrus Logic, Inc. Frequency and direction-dependent ambient sound handling in personal audio devices having adaptive noise cancellation (ANC)
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
US9076427B2 (en) 2012-05-10 2015-07-07 Cirrus Logic, Inc. Error-signal content controlled adaptation of secondary and leakage path models in noise-canceling personal audio devices
EP2667379B1 (en) * 2012-05-21 2018-07-25 Harman Becker Automotive Systems GmbH Active noise reduction
CN102769816B (en) * 2012-07-18 2015-05-13 歌尔声学股份有限公司 Device and method for testing noise-reduction earphone
US9532139B1 (en) 2012-09-14 2016-12-27 Cirrus Logic, Inc. Dual-microphone frequency amplitude response self-calibration
US8798283B2 (en) * 2012-11-02 2014-08-05 Bose Corporation Providing ambient naturalness in ANR headphones
US9107010B2 (en) 2013-02-08 2015-08-11 Cirrus Logic, Inc. Ambient noise root mean square (RMS) detector
US9369798B1 (en) 2013-03-12 2016-06-14 Cirrus Logic, Inc. Internal dynamic range control in an adaptive noise cancellation (ANC) system
US9106989B2 (en) 2013-03-13 2015-08-11 Cirrus Logic, Inc. Adaptive-noise canceling (ANC) effectiveness estimation and correction in a personal audio device
US9414150B2 (en) 2013-03-14 2016-08-09 Cirrus Logic, Inc. Low-latency multi-driver adaptive noise canceling (ANC) system for a personal audio device
US9215749B2 (en) 2013-03-14 2015-12-15 Cirrus Logic, Inc. Reducing an acoustic intensity vector with adaptive noise cancellation with two error microphones
US9635480B2 (en) 2013-03-15 2017-04-25 Cirrus Logic, Inc. Speaker impedance monitoring
US9502020B1 (en) 2013-03-15 2016-11-22 Cirrus Logic, Inc. Robust adaptive noise canceling (ANC) in a personal audio device
US9467776B2 (en) 2013-03-15 2016-10-11 Cirrus Logic, Inc. Monitoring of speaker impedance to detect pressure applied between mobile device and ear
US9208771B2 (en) 2013-03-15 2015-12-08 Cirrus Logic, Inc. Ambient noise-based adaptation of secondary path adaptive response in noise-canceling personal audio devices
US10206032B2 (en) 2013-04-10 2019-02-12 Cirrus Logic, Inc. Systems and methods for multi-mode adaptive noise cancellation for audio headsets
US9066176B2 (en) 2013-04-15 2015-06-23 Cirrus Logic, Inc. Systems and methods for adaptive noise cancellation including dynamic bias of coefficients of an adaptive noise cancellation system
US9462376B2 (en) 2013-04-16 2016-10-04 Cirrus Logic, Inc. Systems and methods for hybrid adaptive noise cancellation
US9460701B2 (en) 2013-04-17 2016-10-04 Cirrus Logic, Inc. Systems and methods for adaptive noise cancellation by biasing anti-noise level
US9478210B2 (en) 2013-04-17 2016-10-25 Cirrus Logic, Inc. Systems and methods for hybrid adaptive noise cancellation
US9578432B1 (en) 2013-04-24 2017-02-21 Cirrus Logic, Inc. Metric and tool to evaluate secondary path design in adaptive noise cancellation systems
US9264808B2 (en) 2013-06-14 2016-02-16 Cirrus Logic, Inc. Systems and methods for detection and cancellation of narrow-band noise
US9392364B1 (en) 2013-08-15 2016-07-12 Cirrus Logic, Inc. Virtual microphone for adaptive noise cancellation in personal audio devices
US9666176B2 (en) 2013-09-13 2017-05-30 Cirrus Logic, Inc. Systems and methods for adaptive noise cancellation by adaptively shaping internal white noise to train a secondary path
US9620101B1 (en) 2013-10-08 2017-04-11 Cirrus Logic, Inc. Systems and methods for maintaining playback fidelity in an audio system with adaptive noise cancellation
US9704472B2 (en) 2013-12-10 2017-07-11 Cirrus Logic, Inc. Systems and methods for sharing secondary path information between audio channels in an adaptive noise cancellation system
US10219071B2 (en) 2013-12-10 2019-02-26 Cirrus Logic, Inc. Systems and methods for bandlimiting anti-noise in personal audio devices having adaptive noise cancellation
US10382864B2 (en) 2013-12-10 2019-08-13 Cirrus Logic, Inc. Systems and methods for providing adaptive playback equalization in an audio device
US9369557B2 (en) 2014-03-05 2016-06-14 Cirrus Logic, Inc. Frequency-dependent sidetone calibration
US9479860B2 (en) 2014-03-07 2016-10-25 Cirrus Logic, Inc. Systems and methods for enhancing performance of audio transducer based on detection of transducer status
US9648410B1 (en) 2014-03-12 2017-05-09 Cirrus Logic, Inc. Control of audio output of headphone earbuds based on the environment around the headphone earbuds
US9319784B2 (en) 2014-04-14 2016-04-19 Cirrus Logic, Inc. Frequency-shaped noise-based adaptation of secondary path adaptive response in noise-canceling personal audio devices
EP3152756B1 (en) * 2014-06-09 2019-10-23 Dolby Laboratories Licensing Corporation Noise level estimation
US9609416B2 (en) 2014-06-09 2017-03-28 Cirrus Logic, Inc. Headphone responsive to optical signaling
US10181315B2 (en) 2014-06-13 2019-01-15 Cirrus Logic, Inc. Systems and methods for selectively enabling and disabling adaptation of an adaptive noise cancellation system
WO2016029461A1 (en) * 2014-08-29 2016-03-03 安百特半导体有限公司 Feedforward-and-feedback-combined noise cancellation earphone and drive circuit thereof
US9478212B1 (en) 2014-09-03 2016-10-25 Cirrus Logic, Inc. Systems and methods for use of adaptive secondary path estimate to control equalization in an audio device
US9552805B2 (en) 2014-12-19 2017-01-24 Cirrus Logic, Inc. Systems and methods for performance and stability control for feedback adaptive noise cancellation
CN104602163B (en) * 2014-12-31 2017-12-01 歌尔股份有限公司 Active noise reduction earphone and method for noise reduction control and system applied to the earphone
WO2017029550A1 (en) 2015-08-20 2017-02-23 Cirrus Logic International Semiconductor Ltd Feedback adaptive noise cancellation (anc) controller and method having a feedback response partially provided by a fixed-response filter
US9578415B1 (en) 2015-08-21 2017-02-21 Cirrus Logic, Inc. Hybrid adaptive noise cancellation system with filtered error microphone signal
US9773491B2 (en) * 2015-09-16 2017-09-26 Bose Corporation Estimating secondary path magnitude in active noise control
US9923550B2 (en) 2015-09-16 2018-03-20 Bose Corporation Estimating secondary path phase in active noise control
US10013966B2 (en) 2016-03-15 2018-07-03 Cirrus Logic, Inc. Systems and methods for adaptive active noise cancellation for multiple-driver personal audio device
US9860626B2 (en) 2016-05-18 2018-01-02 Bose Corporation On/off head detection of personal acoustic device
US11445306B2 (en) 2016-08-26 2022-09-13 Starkey Laboratories, Inc. Method and apparatus for robust acoustic feedback cancellation
US9838812B1 (en) 2016-11-03 2017-12-05 Bose Corporation On/off head detection of personal acoustic device using an earpiece microphone
US11030989B2 (en) 2016-12-22 2021-06-08 Synaptics Incorporated Methods and systems for end-user tuning of an active noise cancelling audio device
US10751524B2 (en) * 2017-06-15 2020-08-25 Cochlear Limited Interference suppression in tissue-stimulating prostheses
US10235987B1 (en) * 2018-02-23 2019-03-19 GM Global Technology Operations LLC Method and apparatus that cancel component noise using feedforward information
US11681001B2 (en) 2018-03-09 2023-06-20 The Board Of Trustees Of The Leland Stanford Junior University Deep learning method for nonstationary image artifact correction
US10951974B2 (en) 2019-02-14 2021-03-16 David Clark Company Incorporated Apparatus and method for automatic shutoff of aviation headsets
GB2582373B (en) 2019-03-22 2021-08-11 Dyson Technology Ltd Noise control
GB2582374B (en) * 2019-03-22 2021-08-18 Dyson Technology Ltd Noise control
CN111800723B (en) * 2019-06-19 2021-07-23 深圳市豪恩声学股份有限公司 Active noise reduction earphone testing method and device, terminal equipment and storage medium
CN110262243B (en) * 2019-07-02 2022-12-23 上海大学 Micro-vibration active control method for hybrid parallel adaptive structure
US10764699B1 (en) 2019-08-09 2020-09-01 Bose Corporation Managing characteristics of earpieces using controlled calibration
US11404040B1 (en) 2019-12-19 2022-08-02 Dialog Semiconductor B.V. Tools and methods for designing feedforward filters for use in active noise cancelling systems
US10937410B1 (en) * 2020-04-24 2021-03-02 Bose Corporation Managing characteristics of active noise reduction
KR102293882B1 (en) * 2021-05-21 2021-08-25 국방과학연구소 Apparatus and method for reducing noise of own ship in a towed line array sonar system
US11678116B1 (en) 2021-05-28 2023-06-13 Dialog Semiconductor B.V. Optimization of a hybrid active noise cancellation system
US11942068B2 (en) * 2022-03-17 2024-03-26 Airoha Technology Corp. Adaptive active noise control system with unstable state handling and associated method
US11875772B2 (en) * 2022-03-17 2024-01-16 Airoha Technology Corp. Adaptive active noise control system with double talk handling and associated method

Family Cites Families (2)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US6396930B1 (en) * 1998-02-20 2002-05-28 Michael Allen Vaudrey Active noise reduction for audiometry
US6741707B2 (en) * 2001-06-22 2004-05-25 Trustees Of Dartmouth College Method for tuning an adaptive leaky LMS filter

Non-Patent Citations (1)

* Cited by examiner, † Cited by third party
Title
See references of EP1744713A4 *

Cited By (24)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
JP2007241104A (en) * 2006-03-10 2007-09-20 Saitama Univ Adaptive linear prediction unit, speech enhancing device, and speech enhancing system
JP2008116782A (en) * 2006-11-07 2008-05-22 Sony Corp Noise canceling system and noise canceling method
KR101357935B1 (en) 2006-11-07 2014-02-03 소니 주식회사 Noise canceling system and noise canceling method
US8401205B2 (en) 2006-11-07 2013-03-19 Sony Corporation Noise canceling system and noise canceling method
GB2445388A (en) * 2007-02-16 2008-07-09 Sonaptic Ltd Noise-canceling ear-worn speaker devices with vents to bypass seals
GB2445388B (en) * 2007-02-16 2009-01-07 Sonaptic Ltd Ear-worn speaker-carrying devices
JP2008250270A (en) * 2007-03-02 2008-10-16 Sony Corp Signal processing apparatus and signal processing method
JP5114611B2 (en) * 2007-09-28 2013-01-09 株式会社DiMAGIC Corporation Noise control system
US8693699B2 (en) 2008-07-29 2014-04-08 Dolby Laboratories Licensing Corporation Method for adaptive control and equalization of electroacoustic channels
US8355513B2 (en) 2009-04-28 2013-01-15 Burge Benjamin D Convertible filter
WO2010129241A1 (en) * 2009-04-28 2010-11-11 Bose Corporation Dynamically configurable anr filter and signal processing topology
US8165313B2 (en) 2009-04-28 2012-04-24 Bose Corporation ANR settings triple-buffering
US8090114B2 (en) 2009-04-28 2012-01-03 Bose Corporation Convertible filter
US8184822B2 (en) 2009-04-28 2012-05-22 Bose Corporation ANR signal processing topology
US8073150B2 (en) 2009-04-28 2011-12-06 Bose Corporation Dynamically configurable ANR signal processing topology
US8073151B2 (en) 2009-04-28 2011-12-06 Bose Corporation Dynamically configurable ANR filter block topology
EP2597889A1 (en) 2011-11-22 2013-05-29 Parrot Headphones with non-adaptive active noise control
JP2013037381A (en) * 2012-10-17 2013-02-21 Sony Corp Noise canceling device and noise canceling method
CN105785349A (en) * 2016-05-09 2016-07-20 浙江大学 Noise removing method for phased-array three dimensional acoustic photographing sonar
CN109697986A (en) * 2018-09-19 2019-04-30 四川大学 A kind of adaptive deviation compensation echo cancel method based on minimum cube absolute value
CN109697986B (en) * 2018-09-19 2020-12-18 四川大学 Adaptive bias compensation echo cancellation method based on minimum cubic absolute value
WO2020241915A1 (en) * 2019-05-29 2020-12-03 이대훈 Bone conduction bluetooth mono earset
CN110610693A (en) * 2019-08-09 2019-12-24 漳州立达信光电子科技有限公司 Weighted mixed type active anti-noise system and controller
CN110610693B (en) * 2019-08-09 2022-04-05 漳州立达信光电子科技有限公司 Weighted mixed type active anti-noise system and controller

Also Published As

Publication number Publication date
JP2007536877A (en) 2007-12-13
US20040264706A1 (en) 2004-12-30
EP1744713A2 (en) 2007-01-24
KR20070010166A (en) 2007-01-22
WO2005112849A3 (en) 2006-01-12
US6996241B2 (en) 2006-02-07
EP1744713A4 (en) 2008-07-30

Similar Documents

Publication Publication Date Title
WO2005112849A2 (en) Tuned feedforward lms filter with feedback control
US6741707B2 (en) Method for tuning an adaptive leaky LMS filter
RU2545384C2 (en) Active suppression of audio noise
Zelinski A microphone array with adaptive post-filtering for noise reduction in reverberant rooms
US9704471B1 (en) Adaptive modeling of secondary path in an active noise control system
EP1439736A1 (en) Feedback cancellation device
CN106161751B (en) A kind of noise suppressing method and device
US8687819B2 (en) Method for monitoring the influence of ambient noise on stochastic gradient algorithms during identification of linear time-invariant systems
Aslam Maximum likelihood least squares identification method for active noise control systems with autoregressive moving average noise
Chien et al. Affine-projection-like maximum correntropy criteria algorithm for robust active noise control
Cartes et al. Experimental evaluation of leaky least-mean-square algorithms for active noise reduction in communication headsets
Zhang et al. On comparison of online secondary path modeling methods with auxiliary noise
Fabry et al. Active noise control with reduced-complexity Kalman filter
Vicen-Bueno et al. Modified LMS-based feedback-reduction subsystems in digital hearing aids based on WOLA filter bank
Roy et al. Active noise control using filtered-xLMS and feedback ANC filter algorithms
US20230362542A1 (en) Audio controller for a semi-adaptive active noise reduction device
Akhtar et al. Automatic tuning of probe noise for continuous acoustic feedback cancelation in hearing aids
WO2021016000A2 (en) Frequency domain adaptation with dynamic step size adjustment based on analysis of statistic of adaptive filter coefficient movement
Cartes et al. Lyapunov tuning of the leaky LMS algorithm for single-source, single-point noise cancellation
US11984107B2 (en) Audio signal processing method and system for echo suppression using an MMSE-LSA estimator
Guldenschuh Secondary-path models in adaptive-noise-control headphones
Huang et al. Directional dependency for feedforward active noise control systems with in-ear headphones
Lu et al. A survey on active noise control techniques--Part I: Linear systems
Lee et al. Channel prediction-based noise reduction algorithm for dual-microphone mobile phones
Niedźwiecki et al. Hybrid SONIC: joint feedforward–feedback narrowband interference canceler

Legal Events

Date Code Title Description
AK Designated states

Kind code of ref document: A2

Designated state(s): AE AG AL AM AT AU AZ BA BB BG BR BW BY BZ CA CH CN CO CR CU CZ DE DK DM DZ EC EE EG ES FI GB GD GE GH GM HR HU ID IL IN IS JP KE KG KM KP KR KZ LC LK LR LS LT LU LV MA MD MG MK MN MW MX MZ NA NI NO NZ OM PG PH PL PT RO RU SC SD SE SG SK SL SM SY TJ TM TN TR TT TZ UA UG US UZ VC VN YU ZA ZM ZW

AL Designated countries for regional patents

Kind code of ref document: A2

Designated state(s): BW GH GM KE LS MW MZ NA SD SL SZ TZ UG ZM ZW AM AZ BY KG KZ MD RU TJ TM AT BE BG CH CY CZ DE DK EE ES FI FR GB GR HU IE IS IT LT LU MC NL PL PT RO SE SI SK TR BF BJ CF CG CI CM GA GN GQ GW ML MR NE SN TD TG

121 Ep: the epo has been informed by wipo that ep was designated in this application
DPEN Request for preliminary examination filed prior to expiration of 19th month from priority date (pct application filed from 20040101)
WWE Wipo information: entry into national phase

Ref document number: 2005758737

Country of ref document: EP

WWE Wipo information: entry into national phase

Ref document number: 1020067023356

Country of ref document: KR

WWE Wipo information: entry into national phase

Ref document number: 2007513154

Country of ref document: JP

NENP Non-entry into the national phase

Ref country code: DE

WWW Wipo information: withdrawn in national office

Country of ref document: DE

WWP Wipo information: published in national office

Ref document number: 1020067023356

Country of ref document: KR

WWP Wipo information: published in national office

Ref document number: 2005758737

Country of ref document: EP