CN111916099A - Adaptive echo cancellation device and method for variable-step hearing aid - Google Patents
Adaptive echo cancellation device and method for variable-step hearing aid Download PDFInfo
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Abstract
The invention discloses a self-adaptive echo cancellation device and a self-adaptive echo cancellation method for a variable-step-size hearing aid, wherein the self-adaptive echo cancellation device comprises a single-frequency tone detector, a step-size controller and a self-adaptive filter: the single-frequency tone detector is used for carrying out spectrum energy analysis on the error signal sample, calculating the current state parameter of the hearing aid system and transmitting the parameter to the step length controller; the step length controller is used for judging the state of the system according to the system state parameters and the normalized error mean value obtained from the single-frequency tone detector, calculating the time-varying step length parameters of the self-adaptive filter and transmitting the time-varying step length parameters to the self-adaptive filter; the self-adaptive filter is used for filtering the cached far-end signal sample of the loudspeaker, calculating and estimating an echo signal and then outputting the echo signal, and iteratively updating the self-adaptive filter according to the time-varying step length parameter calculated by the step length controller. The method solves the problems that the self-adaptive filter in the prior art is low in convergence speed and high in algorithm complexity and is difficult to realize, and is suitable for a digital hearing aid system.
Description
Technical Field
The invention relates to a step-size-variable hearing aid self-adaptive echo cancellation device and a step-size-variable hearing aid self-adaptive echo cancellation method.
Background
With the development of industrialization and the increasing aging of recent years, the number of people suffering from hearing loss diseases is continuously increasing, and the selection of hearing aids to improve hearing loss is the most common and quick auxiliary means. However, the hearing aid has a very easy structure and function to amplify echo and generate howling, so the echo cancellation algorithm is one of the key algorithms of the digital hearing aid, which estimates the echo signal by estimating the characteristic parameters of the echo path, using the far-end signal of the loudspeaker as a reference, and then estimates the echo signal from the received near-end microphone signal. Since the echo path is usually unknown and time-varying, especially since slight wear adjustments during use of the hearing aid change the echo path, the adaptive filter becomes a key component of the hearing aid echo cancellation module by virtue of its ability to track the time-varying system.
The patent with publication number CN101179294B discloses an adaptive echo canceller and an echo cancellation method thereof, and as described in the background of paragraph [0011] of the present application, "in an embedded system, an adaptive algorithm commonly used in an echo canceller is an algorithm group based on the steepest descent method. A representation of such an adaptive algorithm is: LMS (least mean square error) algorithm whose minimization criterion is the root mean square error. The self-adaptive algorithm has the advantages of small calculation amount, strong robustness and easy realization, and is widely adopted in practice. The disadvantages are: the convergence speed is slow and the convergence performance is sensitive to energy variations of the input signal. The NLMS (energy normalized least mean square error) algorithm is an improved algorithm of the LMS algorithm and overcomes the defect that the LMS algorithm is sensitive to the energy of an input signal. The NLMS algorithm and its various modifications are the adaptive filtering algorithm in echo cancellers that are mainly used at present.
Taking an adaptive algorithm LMS as an example, in the adaptive process of the filter, in practical situations, especially in the field of hearing aids, noise or voice signals inevitably exist in the near-end signal except for the desired signal, and the noise equivalent to the noise added with a large signal greatly affects the convergence rate of the adaptive process and may cause divergence in severe cases, and this condition is called double-ended sounding (DT) in the hearing aid echo cancellation system. The misadjustment of the self-adaptation process of the filter can be slowed down to a certain extent by directly reducing the convergence step size parameter, however, the convergence speed of the algorithm and the tracking speed of the time-varying system can be seriously reduced, and the calculation difficulty in the fixed-point system can be caused by the too small step size parameter.
Some echo cancellation algorithms include a double talk detection module (DTD): the double talk detection is to determine the talk state of the current system (near-end talk, far-end talk, double-end talk), when the system is determined to be double-end talk, the adaptive filter does not update the coefficient, and the general DTD method has a disadvantage that the method cannot adapt to the characteristic that the echo path in the echo cancellation environment is changed continuously. Therefore, many scholars have actively studied the existing problems in the adaptive filter echo cancellation method and have proposed many improvements. The method mainly comprises the steps of giving a variable step function or analyzing an input end signal and the like to control the updating of the coefficient of the adaptive filter so as to improve the stable initial convergence speed and the stable error of an echo path. However, these methods generally have the problems of high requirement on computational accuracy and high computational complexity, and are difficult to implement in a hearing aid echo cancellation system, and in the prior art, there is no method suitable for a hearing aid and considering power consumption and computational complexity, which can solve the above problems well.
Disclosure of Invention
Aiming at the problems, the invention provides a variable step length hearing aid self-adaptive echo cancellation device and an echo cancellation method, which solve the problems that the self-adaptive filter in the prior art is low in convergence speed and too high in algorithm complexity and difficult to realize; furthermore, the problems that the steady-state error of the self-adaptive filter is large, double-end sounding and parameter misadjustment can be caused by related data in the prior art are solved.
In order to achieve the technical purpose and achieve the technical effect, the invention is realized by the following technical scheme:
a step size-variable hearing aid adaptive echo cancellation device comprises a single-frequency tone detector, a step size controller and an adaptive filter:
the single tone detector is used for correcting error signalsCarrying out spectrum energy analysis on the sample, calculating the current state parameter of the hearing aid system, and transmitting the parameter to the step length controller;
the step size controller is used for obtaining a normalized error mean value according to the system state parameters obtained from the single-frequency tone detectorJudging the state of the system, and calculating the time-varying step length parameter of the adaptive filterAnd passes it to the adaptive filter;
the adaptive filter is used for buffering loudspeaker far-end signalsFiltering the samples to calculate an estimated echo signalThen outputting, and carrying out iterative updating on the self-adaptive filter according to the time-varying step length parameter calculated by the step length controller;
Normalized mean of errorIs an error signalRelative to the far-end signalNormalized long term average.
Preferably, the single-frequency tone detector includes a framing FFT module and a spectral energy analysis module:
the frame FFT module is used for receiving the buffered error signalFor the error signalPerforming fast Fourier transform according to a set frequency, calculating a power spectrum of the signal and transmitting the power spectrum to a spectrum energy analysis module;
the spectrum energy analysis module analyzes the maximum single-frequency energy of the obtained signal spectrumRatio of maximum single-frequency energy to total energy of power spectrumAnd the obtained system state parameters are transmitted to the step size controller.
Preferably, the step length controller comprises a state selection module, and the state selection module selects the state of the system according to the system state parameters obtained by the single-frequency tone detector, and controls and adjusts the time-varying step length parametersTime-varying step size parameterTo an adaptive filter.
Preferably, the step size controller corrects the coefficient according to the normalizationAnd most obtained by single tone detectorLarge single frequency energyAnd the ratio of maximum single-frequency energy to total energy of power spectrumJudging the threshold value, and controlling and adjusting the time-varying step length parameter according to the judgment resultThe method specifically comprises the following steps:
1) if the maximum single-frequency energy accounts for the total energy ratio of the power spectrumLess than a set thresholdIf yes, the system is judged to be in steady state convergence, and a preset steady state step length parameter is adoptedAs a time-varying step size parameter;
2) if the maximum single-frequency energy accounts for the total energy ratio of the power spectrumGreater than a set thresholdAnd maximum single frequency energyLess than a set thresholdIf the system is determined to be in a related interference state, the normalization correction coefficient is usedCorrecting time-varying step size parameters:;
3) If the maximum single-frequency energy accounts for the total energy ratio of the power spectrumGreater than a set thresholdAnd maximum single frequency energyGreater than a set thresholdAnd normalized correction coefficientLess than a set thresholdIf yes, the system is judged to be in a howling state, and at the moment, howling is preferentially processed, and a fixed step length parameter is adoptedFilter iteration is accelerated:wherein;
4) if the maximum single-frequency energy accounts for the total energy ratio of the power spectrumGreater than a set thresholdAnd maximum single frequency energy Greater than a set thresholdAnd normalized correction coefficientGreater than a set thresholdIf the system is in a double-end sounding state, the time-varying step length parameter is set to zero to stop the adaptive iteration of the filter。
Preferably, the adaptive filter includes a normalized error signal processing module, a coefficient updating module, and a FIR filter module:
the normalized error signal processing module is used for processing the error signalFar-end signal relative to loudspeakerIs normalized by the square of the Euclidean norm to obtain a normalized error signal;
Wherein,is the far-end signal vector for the corresponding time delay,in order to adapt the order of the filter,is a constant;
normalizing error signals according to set frequency by convex combination first-order recursion processPerforming snapshot smoothing to obtain a normalized average error;
the coefficient updating module obtains time-varying step length parameters according to the normalized error signals, the normalized average errors and the step length controllerIteratively updating the adaptive filter coefficients;
the FIR filter module is used for far-end signalFiltering to obtain estimated echo signalAnd then outputting.
A variable step size hearing aid adaptive echo cancellation device according to any one of the preceding claims, adapted for use in a digital hearing aid system.
Correspondingly, the adaptive echo cancellation method for the variable-step hearing aid comprises the following steps:
A. according to the cached far-end signal sample, the estimated echo signal is obtained through filtering of a self-adaptive FIR filter: Wherein,is an FIR filter systemThe number of the first and second groups is,is composed ofThe conjugate transpose of (1); and the sampled near-end voice signalSubtracting the estimated echo signalTo obtain an error signal;
B. Performing time-frequency transformation on the error signal, calculating the power spectrum of the error signal and analyzing the energy distribution characteristic to obtain system state parameters;
C. judging the state of the system according to the system state parameters, and controlling the self-adaptive time-varying step length parameters;
D. For error signalFar-end signal relative to loudspeakerThe square of the Euclidean norm is normalized to obtain a normalized error mean valueCarrying out smooth processing on the maximum value of each frame of the normalized error signal by using a convex combination first-order recursion process, and using the maximum value as an upper limit threshold value of the normalized adaptive error;
E. using modified time-varying step-size parametersAnd the normalized error iteratively updates the FIR filter coefficient:。
preferably, the step B specifically includes:
b1, continuously filling error signals into a buffer area of the single-frequency tone detector, performing one-time fast Fourier transform operation after the buffer area is full of data, resetting the buffer to obtain the frequency spectrum of the signals, and further calculating the power spectrum of the signals:
Whereinas the number of frames,are the frequency points of the frequency,;is FFT operation;the number of fast Fourier transform points;
b2, finding the maximum energy frequency point according to the power spectrum of the signalCalculating the energy sum of the maximum energy frequency point and the adjacent frequency points:
Calculating the ratio of the maximum single-frequency energy to the total energy of the power spectrum:
Preferably, the step C specifically includes:
c1 maximum ratio of single-frequency energy to total energy of power spectrum obtained by single-frequency tone detectorSmooth normalization correction coefficient:
c2 maximum single frequency energyRatio of maximum single-frequency energy to total energy of power spectrumNormalized correction factorAnd judging a threshold value, and adjusting a time-varying step length parameter according to the judgment results of the three state parameters, specifically:
1) if the maximum single-frequency energy accounts for the total energy ratio of the power spectrumLess than a set thresholdIf yes, the system is judged to be in steady state convergence, and a preset steady state step length parameter is adoptedAs a time-varying step size parameter;
2) if the maximum single-frequency energy accounts for the total energy ratio of the power spectrumGreater than a set thresholdAnd maximum single frequency energyLess than a set thresholdIf the system is determined to be in a related interference state, the normalization correction coefficient is usedCorrecting time-varying step length parameters:;
3) if the maximum single-frequency energy accounts for the total energy ratio of the power spectrumGreater than a set thresholdAnd maximum single frequency energyGreater than a set thresholdAnd normalized correction coefficientLess than a set thresholdIf yes, the system is judged to be in a howling state, and at the moment, howling is preferentially processed, and a fixed step length parameter is adoptedFilter iteration is accelerated:wherein;
4) if the maximum single-frequency energy accounts for the total energy ratio of the power spectrumGreater than a set thresholdAnd maximum single frequency energyGreater than a set thresholdAnd normalized correction coefficientGreater than a set thresholdIf so, judging that the system is in a double-end sounding state, and setting the time-varying step length parameter to zero at the moment to stop the adaptive iteration of the filter:。
preferably, the step D specifically includes:
d1, for the current error signalFar-end signal relative to loudspeakerSquare of the euclidean norm ofNormalization is performed to obtain a normalized error signal:
d2, calculating the absolute value of the normalized error signal of each frameMaximum value ofAnd smoothing the data by convex combination first order recursion process to obtain normalized error mean of normalized error signal:
d3, normalizing the normalized error mean of the error signalAs the upper threshold limit for the error signal.
The invention has the beneficial effects that:
1) the invention takes the detection data of the single-frequency tone detector as the basis, and by means of the multiplier, FFT and other modules of the hearing aid chip, the algorithm is low in complexity and power consumption, is easy to realize in the digital hearing aid, and solves the problems that the self-adaptive filter in the prior art is low in convergence speed and high in algorithm complexity and is difficult to realize.
2) According to the method, the iteration step length is corrected according to the distribution characteristics of the frequency spectrum energy of the historical data after echo elimination to weaken the maladjustment problem caused by single-frequency noise, the working state of the adaptive filter is switched to quickly inhibit howling, good balance is achieved in the convergence speed and the steady-state error of the adaptive filter, and a time-varying system can be quickly tracked and howling inhibition can be quickly achieved.
3) The invention takes the long-term average value of the normalized error signal as the dynamic threshold value to control the iteration step length of the filter not to have sudden change, weakens the influence of double-end talkback under the conditions of higher robustness and faster convergence speed, and solves the problems of large steady-state error of the self-adaptive filter, double-end sounding, parameter imbalance generation of related data and the like in the prior art.
4) The adaptive filter has strong anti-interference capability in an external noise or double-end talkback scene, and can obviously prevent the iterative divergence of the adaptive filter. When the device is in an environment with strong noise single-frequency energy, the detuning can be effectively prevented.
Drawings
Fig. 1 is a block diagram of an adaptive echo cancellation device for a hearing aid with variable step size according to the present invention;
fig. 2 is a diagram comparing the convergence performance of the present method and the conventional method in the presence of external interference.
Detailed Description
The present invention will be better understood and implemented by those skilled in the art by the following detailed description of the technical solution of the present invention with reference to the accompanying drawings and specific examples, which are not intended to limit the present invention.
As shown in fig. 1, a step-size-variable hearing aid adaptive echo cancellation apparatus mainly includes a single-frequency tone detector, a step-size controller, and an adaptive filter, and specifically, the following components are mainly introduced:
the single tone detector is used for correcting error signalsAnd carrying out spectrum energy analysis on the sample, calculating the current state parameter of the hearing aid system, and transmitting the parameter to the step size controller.
Adaptive echo cancellation device for external input signalCollecting to obtain near-end voice signalWherein the error signalFor near-end speech signalsSubtracting the estimated echo signal。
The step size controller is used for obtaining a normalized error mean value according to the system state parameters obtained from the single-frequency tone detectorJudging the state of the system, and calculating the time-varying step length parameter of the adaptive filterAnd passes it to the adaptive filter. Wherein the error mean is normalizedIs an error signalRelative to the far-end signalNormalized long term average.
The adaptive filter is used for buffering loudspeaker far-end signalsFiltering the samples to calculate an estimated echo signalAnd then outputting, and carrying out iterative updating on the self-adaptive filter according to the time-varying step length parameter calculated by the step length controller. In the filtering process, the time-varying step size parameter updated last time is adopted for filtering, and the time-varying step size parameter updated this time is used for the next filtering process, so that the operation can be reduced, the power consumption of the hearing aid is reduced, and the implementation in the digital hearing aid is easier.
Preferably, the main structure and function of each component of the adaptive echo cancellation device are described as follows:
preferably, the single-frequency tone detector includes a framing FFT module and a spectral energy analysis module, wherein:
the frame FFT module is used for receiving the buffered error signalFor the error signalPerforming Fast Fourier Transform (FFT) according to a set frequency, calculating a power spectrum of the signal and transmitting the power spectrum to a spectrum energy analysis module;
the spectrum energy analysis module analyzes the maximum single-frequency energy of the obtained signal spectrumMaximum single frequency energyRatio of quantity to total energy of power spectrumAnd the obtained system state parameters are transmitted to the step size controller.
Preferably, the step length controller comprises a state selection module, and the state selection module selects the state of the system according to the system state parameters obtained by the single-frequency tone detector, and controls and adjusts the time-varying step length parametersTime-varying step size parameterTo an adaptive filter.
For example, the step size controller corrects the coefficient according to the normalization factorAnd maximum single frequency energy obtained by the single frequency tone detectorAnd the ratio of maximum single-frequency energy to total energy of power spectrumJudging the threshold value, and controlling and adjusting the time-varying step length parameter according to the judgment resultThe method specifically comprises the following steps:
1) if the maximum single-frequency energy accounts for the total energy ratio of the power spectrumLess than a set thresholdIf yes, the system is judged to be in steady state convergence, and a preset steady state step length parameter is adoptedAs the time-varying step size parameter, in general,the value may be 0.05;
2) if the maximum single-frequency energy accounts for the total energy ratio of the power spectrumGreater than a set thresholdAnd maximum single frequency energyLess than a set thresholdIf the system is determined to be in a related interference state, the normalization correction coefficient is usedCorrecting time-varying step length parameters:;
3) if the maximum single-frequency energy accounts for the total energy ratio of the power spectrumGreater than a set thresholdAnd maximum single frequency energyGreater than a set thresholdAnd normalized correction coefficientLess than a set thresholdIf yes, the system is judged to be in a squeaking state, at the moment, squeaking is preferentially processed, and a larger fixed step length parameter is adoptedFilter iteration is accelerated:wherein;
4) if the maximum single-frequency energy accounts for the total energy ratio of the power spectrumGreater than a set thresholdAnd maximum single frequency energyGreater than a set thresholdAnd normalized correction coefficientGreater than a set thresholdIf so, judging that the system is in a double-end sounding state, and setting the time-varying step length parameter to zero at the moment to stop the adaptive iteration of the filter:。
wherein the set threshold value 、、 Are all constants that can be derived from the particular system state adjustments.
Preferably, the adaptive filter includes a normalized error signal processing module, a coefficient updating module, and a FIR filter module, wherein:
the normalized error signal processing module is used for processing the error signalFar-end signal relative to loudspeakerIs normalized by the square of the Euclidean norm to obtain a normalized error signal;
Wherein,is the remote signal vector corresponding to the time delay, D is the adaptive filter order,is a very small constant and is used to prevent the denominator from being too small to cause divergence.
Normalizing error signals according to set frequency by convex combination first-order recursion processAnd performing snapshot smoothing to obtain a normalized average error.
The coefficient isThe new module obtains time-varying step length parameters according to the normalized error signal, the normalized average error and the step length controllerAnd iteratively updating the adaptive filter coefficients.
The FIR filter module is used for far-end signalFiltering to obtain estimated echo signalAnd then outputting.
Correspondingly, the hearing aid adaptive echo cancellation device according to any one of the above items is suitable for a digital hearing aid system. The digital hearing aid automatically collects the conditions of the acoustic signal type, the signal-to-noise ratio, the strength difference of the front microphone and the rear microphone and the like of the environment where the digital hearing aid is located, defines different environments, and automatically adjusts the characteristics of noise reduction, direction, compression ratio and the like so as to adapt to the continuously changing environment. The phenomenon that a user of the analog machine cannot hear the sound with small sound and is difficult to hear with loud sound is avoided. Generally, a digital hearing aid system mainly includes an echo cancellation device, a microphone (or microphone), an amplifier, a receiver (or earphone), a battery, electro-acoustic devices such as various volume or tone control knobs, and a housing, wherein the echo cancellation device can adopt any one of the above-mentioned step-size-variable hearing aid adaptive echo cancellation devices.
Correspondingly, the adaptive echo cancellation method for the variable-step hearing aid comprises the following steps:
A. according to the cached far-end signal sample, the estimated echo signal is obtained through filtering of a self-adaptive FIR filter: Wherein, for the coefficients of the FIR filter,is composed ofThe conjugate transpose of (1); and the sampled near-end voice signalSubtracting the estimated echo signalTo obtain an error signal。
B. Performing time-frequency transformation on the error signal, calculating a power spectrum of the error signal, analyzing energy distribution characteristics, and obtaining system state parameters, preferably, the step B specifically includes:
b1, continuously filling error signals into a buffer area of the single-frequency tone detector, performing one-time fast Fourier transform operation after the buffer area is full of data, resetting the buffer to obtain the frequency spectrum of the signals, and further calculating the power spectrum of the signals:
Whereinas the number of frames,are the frequency points of the frequency,;is FFT operation;the number of fast Fourier transform points;
b2, finding the maximum energy frequency point according to the power spectrum of the signalCalculating the energy sum of the maximum energy frequency point and the adjacent frequency points:
Calculating the ratio of the maximum single-frequency energy to the total energy of the power spectrum:
C、Judging the state of the system according to the system state parameters, and controlling the self-adaptive time-varying step length parameters. Preferably, the step C specifically includes:
c1 maximum ratio of single-frequency energy to total energy of power spectrum obtained by single-frequency tone detectorSmooth normalization correction coefficient:
,for controlling correction coefficientBetween 0 and 1;for the set threshold, the value is generally 0.5.
C2 maximum single frequency energyRatio of maximum single-frequency energy to total energy of power spectrumNormalized correction factorPerforming threshold judgment, and adjusting the time-varying step length parameter according to the judgment results of the three state parameters, wherein the preferable step C2 specifically comprises:
1) if the maximum single-frequency energy accounts for the total energy ratio of the power spectrumLess than a set thresholdIf yes, the system is judged to be in steady state convergence, and a preset steady state step length parameter is adoptedAs a time-varying step size parameter;
2) if the maximum single-frequency energy accounts for the total energy ratio of the power spectrumGreater than a set thresholdAnd maximum single frequency energyLess than a set thresholdIf the system is determined to be in a related interference state, the normalization correction coefficient is usedCorrecting time-varying step length parameters:;
3) if the maximum single frequency energy occupies powerRatio of total energy of spectrumGreater than a set thresholdAnd maximum single frequency energyGreater than a set thresholdAnd normalized correction coefficientLess than a set thresholdIf yes, the system is judged to be in a howling state, and at the moment, howling is preferentially processed, and a fixed step length parameter is adoptedFilter iteration is accelerated:wherein;
4) if the maximum single-frequency energy accounts for the total energy ratio of the power spectrumGreater than a set thresholdAnd maximum single frequency energyGreater than a set thresholdAnd normalized correction coefficientGreater than a set thresholdIf so, judging that the system is in a double-end sounding state, and setting the time-varying step length parameter to zero at the moment to stop the adaptive iteration of the filter:。
D. for error signalFar-end signal relative to loudspeakerThe square of the Euclidean norm is normalized to obtain a normalized error mean valueAnd smoothing the maximum value of each frame of the normalized error signal by using a convex combination first-order recursion process to be used as an upper limit threshold value of the normalized adaptive error.
Preferably, the step D specifically includes:
d1, for the current error signalFar-end signal relative to loudspeakerSquare of the euclidean norm ofNormalization is performed to obtain a normalized error signal:
Wherein,is a very small constant and is used to prevent the denominator from being too small to cause divergence.
D2, calculating the absolute value of the normalized error signal of each frameMaximum value ofAnd smoothing the data by convex combination first order recursion process to obtain normalized error mean of normalized error signal:
Wherein,andin order to be a forgetting factor,the value is generally small: preferably, the first and second liquid crystal materials are,,large value, preferably;
D3, normalizing the normalized error mean of the error signalAs the upper limit of the threshold value of the error signal, the sudden change of the error signal caused by the interference situation such as double-end sounding is prevented:
E. using modified time-varying step-size parametersAnd the normalized error iteratively updates the FIR filter coefficient:。
the present invention and the prior art are implemented below in conjunction with the HA320D digital hearing aid chip of Nanjing Tianyue electronics, Inc., where the HA320D digital hearing aid system sample rate is 16kHZ, and the quantization bit number is 16 bits;,,,needs to be measured according to the quantification of the system and various parameter conditions in the howling critical stateObtaining;the value of the carbon dioxide is 0.05,the value is 0.1;the value of the additive is 0.001,the value is 0.5;the value is 0.001.
As shown in fig. 2, in a real hearing aid closed-loop environment, the adaptive echo cancellation device and the echo cancellation method for a hearing aid in variable step sizes of the present invention have the following significant advantages:
1) the anti-interference capability under the external noise or double-end talkback scene is strong, and the iterative divergence of the adaptive filter can be obviously prevented;
2) imbalance can be effectively prevented when the device is in an environment with strong noise single-frequency energy;
3) the convergence speed and the steady-state error of the adaptive filter are well balanced, and a time-varying system and howling suppression can be quickly tracked;
4) by means of the multiplying unit, FFT and other modules of the hearing aid chip, the algorithm is low in complexity and power consumption.
The above description is only a preferred embodiment of the present invention, and not intended to limit the scope of the present invention, and all modifications of equivalent structures and equivalent processes, which are made by using the contents of the present specification and the accompanying drawings, or directly or indirectly applied to other related technical fields, are included in the scope of the present invention.
Claims (10)
1. A step size-variable hearing aid adaptive echo cancellation device is characterized by comprising a single-frequency tone detector, a step size controller and an adaptive filter:
the single tone detector is used for correcting error signalsCarrying out spectrum energy analysis on the sample, calculating the current state parameter of the hearing aid system, and transmitting the parameter to the step length controller;
the step size controller is used for obtaining a normalized error mean value according to the system state parameters obtained from the single-frequency tone detectorJudging the state of the system, and calculating the time-varying step length parameter of the adaptive filterAnd passes it to the adaptive filter;
the adaptive filter is used for buffering loudspeaker far-end signalsFiltering the samples to calculate an estimated echo signalThen outputting, and carrying out iterative updating on the self-adaptive filter according to the time-varying step length parameter calculated by the step length controller;
2. The variable step size hearing aid adaptive echo cancellation device according to claim 1, wherein the single frequency tone detector comprises a frame FFT module and a spectral energy analysis module:
the frame FFT module is used for receiving the buffered error signalFor the error signalPerforming fast Fourier transform according to a set frequency, calculating a power spectrum of the signal and transmitting the power spectrum to a spectrum energy analysis module;
3. The apparatus of claim 1, wherein the step size controller comprises a state selection module, and the state selection module obtains the state according to a single tone detectorSelecting the state of the system by the system state parameters, and controlling and adjusting the time-varying step length parametersTime-varying step size parameterTo an adaptive filter.
4. The adaptive echo cancellation device of claim 3, wherein the step size controller is configured to modify the echo cancellation signal according to a normalization correction factorAnd maximum single frequency energy obtained by the single frequency tone detectorAnd the ratio of maximum single-frequency energy to total energy of power spectrumJudging the threshold value, and controlling and adjusting the time-varying step length parameter according to the judgment resultThe method specifically comprises the following steps:
1) if the maximum single-frequency energy accounts for the total energy ratio of the power spectrumLess than a set thresholdIf yes, the system is judged to be in steady state convergence, and a preset steady state step length parameter is adoptedAs a time-varying step size parameter;
2) if the maximum single-frequency energy accounts for the total energy ratio of the power spectrumGreater than a set thresholdAnd maximum single frequency energyLess than a set thresholdIf the system is determined to be in a related interference state, the normalization correction coefficient is usedCorrecting time-varying step length parameters:;
3) if the maximum single-frequency energy accounts for the total energy ratio of the power spectrumGreater than a set thresholdAnd maximum single frequency energyGreater than a set thresholdAnd normalized correction coefficientLess than a set thresholdIf yes, the system is judged to be in a howling state, and at the moment, howling is preferentially processed, and a fixed step length parameter is adoptedFilter iteration is accelerated:wherein;
4) if the maximum single-frequency energy accounts for the total energy ratio of the power spectrumGreater than a set thresholdAnd maximum single frequency energyGreater than a set thresholdAnd normalized correction coefficientGreater than a set thresholdIf so, judging that the system is in a double-end sounding state, and setting the time-varying step length parameter to zero at the moment to stop the adaptive iteration of the filter:。
5. the variable step size hearing aid adaptive echo cancellation device according to claim 1, wherein the adaptive filter comprises a normalized error signal processing module, a coefficient update module, and a FIR filter module:
the normalized error signal processing module is used for processing the error signalFar-end signal relative to loudspeakerIs normalized by the square of the Euclidean norm to obtain a normalized error signal;
Wherein,is the far-end signal vector for the corresponding time delay,in order to adapt the order of the filter,is a constant;
normalizing error signals according to set frequency by convex combination first-order recursion processPerforming snapshot smoothing to obtain a normalized average error;
the coefficient updating module obtains time-varying step length parameters according to the normalized error signals, the normalized average errors and the step length controllerIteratively updating the adaptive filter coefficients;
6. The variable-stride hearing-aid adaptive echo cancellation device of any one of claims 1-5, wherein the variable-stride hearing-aid adaptive echo cancellation device is adapted for use in a digital hearing-aid system.
7. A method for adaptive echo cancellation in a variable step size hearing aid, comprising the steps of:
A. according to the cached far-end signal sample, the estimated echo signal is obtained through filtering of a self-adaptive FIR filter:Wherein,for the coefficients of the FIR filter,is composed ofThe conjugate transpose of (1); and the sampled near-end voice signalSubtracting the estimated echo signalTo obtain an error signal;
B. Performing time-frequency transformation on the error signal, calculating the power spectrum of the error signal and analyzing the energy distribution characteristic to obtain system state parameters;
C. judging the state of the system according to the system state parameters, and controlling the self-adaptive time-varying step length parameters;
D. For error signalFar-end signal relative to loudspeakerThe square of the Euclidean norm is normalized to obtain a normalized error mean valueCarrying out smooth processing on the maximum value of each frame of the normalized error signal by using a convex combination first-order recursion process, and using the maximum value as an upper limit threshold value of the normalized adaptive error;
8. the method according to claim 7, wherein the step B specifically comprises:
b1, continuously filling error signals into a buffer area of the single-frequency tone detector, performing one-time fast Fourier transform operation after the buffer area is full of data, resetting the buffer to obtain the frequency spectrum of the signals, and further calculating the power spectrum of the signals:
Whereinas the number of frames,are the frequency points of the frequency,;is FFT operation;the number of fast Fourier transform points;
b2, finding the maximum energy frequency point according to the power spectrum of the signalCalculating the energy sum of the maximum energy frequency point and the adjacent frequency points:
Calculating the ratio of the maximum single-frequency energy to the total energy of the power spectrum:
9. The method according to claim 7, wherein the step C specifically comprises:
c1 maximum ratio of single-frequency energy to total energy of power spectrum obtained by single-frequency tone detectorSmooth normalization correction coefficient:
c2 maximum single frequency energyRatio of maximum single-frequency energy to total energy of power spectrumNormalized correction factorAnd judging a threshold value, and adjusting a time-varying step length parameter according to the judgment results of the three state parameters, specifically:
1) if the maximum single-frequency energy accounts for the total energy ratio of the power spectrumLess than a set thresholdIf the system is determined to be stableState convergence by using a preset steady-state step length parameterAs a time-varying step size parameter;
2) if the maximum single-frequency energy accounts for the total energy ratio of the power spectrumGreater than a set thresholdAnd maximum single frequency energyLess than a set thresholdIf the system is determined to be in a related interference state, the normalization correction coefficient is usedCorrecting time-varying step length parameters:;
3) if the maximum single-frequency energy accounts for the total energy ratio of the power spectrumGreater than a set thresholdAnd maximum single frequency energyGreater than a set thresholdAnd normalized correction coefficientLess than a set thresholdIf yes, the system is judged to be in a howling state, and at the moment, howling is preferentially processed, and a fixed step length parameter is adoptedFilter iteration is accelerated:wherein;
4) if the maximum single-frequency energy accounts for the total energy ratio of the power spectrumGreater than a set thresholdAnd maximum single frequency energyGreater than a set thresholdAnd normalized correction coefficientGreater than a set thresholdIf so, judging the system to be in a double-end sounding state, and determining the time-varying step length parameter at the momentZero-stopping the adaptive iteration of the filter:。
10. the method according to claim 7, wherein the step D specifically comprises:
d1, for the current error signalFar-end signal relative to loudspeakerSquare of the euclidean norm ofNormalization is performed to obtain a normalized error signal:
d2, calculating the absolute value of the normalized error signal of each frameMaximum value ofAnd smoothing the data by convex combination first order recursion process to obtain normalized error mean of normalized error signal:
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