CN106161820B - A kind of interchannel decorrelation method for stereo acoustic echo canceler - Google Patents

A kind of interchannel decorrelation method for stereo acoustic echo canceler Download PDF

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CN106161820B
CN106161820B CN201510180915.XA CN201510180915A CN106161820B CN 106161820 B CN106161820 B CN 106161820B CN 201510180915 A CN201510180915 A CN 201510180915A CN 106161820 B CN106161820 B CN 106161820B
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杨鹤飞
郑成诗
李晓东
彭任华
厉剑
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Institute of Acoustics CAS
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Abstract

The present invention provides a kind of interchannel decorrelation methods for stereo acoustic echo canceler, comprising: step 1) distal end microphone picks up acoustic reference signal x1(n) and x2(n);Frequency spectrum X is obtained by Fourier transformation1(k) and X2(k);Step 2) is to frequency spectrum X1(k) and X2(k) high frequency decorrelative transformation is carried out;The time domain reference signal x that obtains that treated1d(n) and x2d(n);Step 3) is to time domain reference signal x1(n) low frequency decorrelative transformation is carried out, three sinusoidal signals are constructed;Above three sinusoidal signal is added to x by time domain addition by step 4)1d(n) in;Obtain x1′(n);x2' (n) is x2d(n);The signal x that step 5) obtains step 4)1' (n) and x2' (n) carries out adaptive channel identification by adaptive algorithm, realizes echo cancellor.The Inter-channel Correlation of boombox signal is effectively reduced in method of the invention, improves the imbalance performance of stereo echo canceller system.

Description

A kind of interchannel decorrelation method for stereo acoustic echo canceler
Technical field
It is the present invention relates to acoustic echo technology for eliminating field, in particular to a kind of for the logical of stereo acoustic echo canceler Decorrelation method between road.
Background technique
In hands-free communication system, due to acoustical signal closed loop transmission and there are acoustic echos.Such issues that in mobile phone It is generally existing in the scenes such as communication, TeleConference Bridge, video conferencing system.Acoustic echo can reduce the communication quality of system, Therefore it needs to be eliminated, generally uses adaptive channel identification algorithm (bibliography [1]: Benesty J, Kellermann K et al.A perspective on stereophonic echo cancellation.Berlin,Heidelberg: Springer-Verlag Berlin Heidelberg, 2011:10.) realize echo elimination.In stereo communication system, There are strong Inter-channel Correlations for boombox signal in near-end room, this leads to the receipts of adaptive channel identification algorithm Hold back that value is not unique, so that the filter for making stereo acoustic echo eliminate system is lacked of proper care larger (bibliography [2]: Benesty J,Morgan D R et al.A better understanding and an improved solution to the specific problems of stereophonic acoustic echo cancellation.IEEE Trans.Speech and Audio Processing,1998,6(2):156-165.)。
In order to improve imbalance performance, need to carry out decorrelation pretreatment to two channel Near end speaker signals in advance;Meanwhile In order to guarantee communication quality, respective handling should not introduce apparent distortion.Have both at home and abroad at present it is some can be used for it is three-dimensional several Learn the interchannel decorrelation method of echo cancellor;Principle is a part of other signals to be added in original signal or directly to original Beginning signal is converted;Wherein most representative is classical non-linear transform algorithm (bibliography [2]), and this method is original The nonlinear transformation ingredient of a part of original signal is added in signal.The defect of these conventional methods is that quality distortion is big, and Decorrelation effect is stronger, causes quality distortion bigger.In order to obtain better compromise between decorrelation effect and quality distortion, In recent years, domestic and international researcher starts with psychologic acoustics effect to carry out interchannel decorrelative transformation, to obtain smaller sound Matter distortion;Auditory masking effect (bibliography [3]: Valin J M.Perceptually-motivated in psychologic acoustics nonlinear channel decorrelation for stereo acoustic echo cancellation,In Proc.Joint Workshop Hands-Free Speech Communication and Microphone Arrays, Trento, May 2008:188-191.) and fundamental frequency loss effect (bibliography [4]: Cecchi S, Romoli L et al.A combined psychoacoustic approach for stereo acoustic echo cancellation.IEEE Trans.Audio, Speech and Language Processing, 2011,19 (6): 1530-1539.) for carrying out low frequency Decorrelative transformation.
Compared with the adaptive channel discrimination method of early stage, the method based on psychologic acoustics effect can be smaller in quality distortion While obtain better decorrelation effect.But for eliminating the more demanding occasion of echo, current is imitated based on psychologic acoustics The method answered is not able to satisfy higher quality distortion requirement, and the calculating process of this method is complex.
Summary of the invention
It is an object of the invention to overcome currently based on the method for psychologic acoustics effect above-mentioned present on elimination echo Defect proposes a kind of interchannel decorrelation method for stereo acoustic echo canceler, and this method can be effectively reduced vertical The Inter-channel Correlation of body sound loudspeaker signal improves the imbalance performance of stereo echo canceller system, it ensure that lesser Quality distortion, and realize that simply computation complexity is low.
In order to achieve the above-mentioned object of the invention, the present invention proposes that a kind of interchannel for stereo acoustic echo canceler goes phase Pass method, which comprises
Step 1) distal end microphone pick-up of acoustic signals obtains stereo reference signal x1(n) and x2(n);Simultaneously to x1(n) And x2(n) Fourier transformation is carried out, frequency spectrum X is obtained1(k) and X2(k);
Step 2) is to frequency spectrum X1(k) and X2(k) high frequency decorrelative transformation is carried out;The time domain reference signal x that obtains that treated1d (n) and x2d(n);
Step 3) is to three-dimensional acoustic reference signal x1(n) low frequency decorrelative transformation is carried out, three sinusoidal signals: s are constructed1 (n)、s2(n) and s3(n);
Step 4) passes through time domain addition for three sinusoidal signal s1(n)、s2(n) and s3(n) it is added to x1d(n) in;? To x '1(n);x′2It (n) is x2d(n);The x '1(n) and x '2It (n) is the stereo signal after Whole frequency band decorrelative transformation;
The signal x ' that step 5) obtains step 4)1(n) and x '2(n) adaptive channel is carried out by adaptive algorithm to distinguish Know, realizes echo cancellor.
In above-mentioned technical proposal, the step 2) specifically:
Simultaneously to frequency spectrum X1(k) and X2(k) high frequency decorrelative transformation is carried out:
Wherein, β (k) is nonlinear weight coefficient, it is contemplated that the relationship of the phase sensitive following frequency of human auditory system, β (k) value are as follows:
Wherein, f (k) is linear frequency value corresponding to discrete frequency indices k,NFFT is in Fu The length of leaf transformation;fsFor the sample frequency of signal;β (k) is in fsIt is maximized at/2;
Judge that analog frequency f (k) corresponding to k is greater than f0It is whether true, the f0It is high frequency or low frequency for division signals Critical value;If a determination be made that negative, 0 is set by Θ value, otherwise, sets Θ value to the value of sinusoidal variations;
To frequency spectrum X1d(k) and X2d(k) inverse Fourier transform is carried out, the time domain reference after high frequency decorrelative transformation is obtained Signal x1d(n) and x2d(n)。
In above-mentioned technical proposal, the f0Value are as follows: f0=1kHz.
In above-mentioned technical proposal, the step 3) is specifically included:
Step 3-1) calculate reference signal x1(n) weighted auto-correlation function ψw(τ);
x1(n) average magnitude difference function φ (τ) is defined as:
φ (τ)=E | x1(n)-x1(n+τ)|} (7)
Wherein, τ is time delay, and desired operation is sought in E { } expression;
x1(n) auto-correlation function is calculated as follows:
ψ (τ)=E { x1(n)x1(n+τ)}. (8)
Weighted auto-correlation function are as follows:
ψw(τ)=ψ (τ)/[φ (τ)+ε], (9)
Wherein ε is the fixed value for preventing quotient from dissipating and introducing, ε=1;
Step 3-2) calculate fundamental frequency cycles estimated value τ0
τ0={ τ: ψw(τ)=max (ψw(τ));fs/350≤τ≤fs/60} (10)
Wherein, max () expression is maximized;
Step 3-3) recalculate fundamental frequency estimated value f0
To with maximum value ψw0) centered on adjacent 3 ψw(τ) value carries out interpolation, then searches again for maximum value and obtains essence Spend higher fundamental frequency cycles estimated value T0, then fundamental frequency value are as follows:
f0=1/T0 (11)
Step 3-4) retrieval reference signal frequency spectrum X1(k), it carries out amplitude weighting and obtains the amplitude A of three low order harmonics ingredientsi And first phase
k0Indicate frequency indices corresponding to fundamental frequency, then two, harmonics corresponds respectively to 2k three times0And 3k0;Utilize three frequencies Rate index value retrieves X1(k), the amplitude and first phase of fundamental frequency and two, harmonics three times are respectively obtained, amplitude weighting is carried out and obtains:
Wherein, A1And σ1For fundamental frequency amplitude, first phase and the amplitude weighting factor, A2And σ2For secondary harmonics amplitude, First phase and the amplitude weighting factor, A3And σ3For for the amplitude of harmonics three times, first phase and the amplitude weighting factor;Angle { } table Show and phase function, σ are taken to plural numberi(i=1,2,3) is in [0,1] interior value, it is preferred that σ1=0.1, σ2=0.2, σ3=0.3;
Step 3-5) three sinusoidal signal s of construction1(n)、s2(n) and s3(n):
Wherein, fi=if0, i=1,2,3.
In above-mentioned technical proposal, the adaptive algorithm are as follows: least fibre method, normalized least fibre method or frequency domain are most Small equal method.
The present invention has the advantages that
1, method of the invention utilizes the nonlinear Phase Modulation of time-varying to traditional nonlinear transformation decorrelation in high band Method improves, thus obtains better high frequency decorrelation effect;
2, method of the invention be based on psychologic acoustics compose advantageous effect, near three low order harmonics ingredients Implantation Energy compared with Weak sinusoidal signal is to carry out low frequency decorrelative transformation, it ensure that lesser quality distortion;
3, method of the invention is injected sinusoidal signal in original signal using time domain addition, and algorithm is made to have lower fortune Calculation amount;
4, method of the invention is lost by adjusting the energy size of injected sinusoidal signal in decorrelation effect and sound quality Different degrees of compromise is carried out between true, therefore can be according to the requirement of concrete application scene by sinusoidal signal in practical application Energy is arranged to different value, such as those occasions of less demanding to sound quality, can increase injected sinusoidal signal Energy is to obtain better decorrelation effect.
Detailed description of the invention
Fig. 1 is that stereo acoustic echo eliminates model general illustration;
Fig. 2 is flow diagram of the present invention for the interchannel decorrelation method of stereo acoustic echo canceler;
Fig. 3 is that the obtained interchannel in example 1 using method and existing non-linear transform algorithm of the invention is relevant Index contrast figure;
Fig. 4 is in example 1 using the obtained imbalance performance of method and existing non-linear transform algorithm of the invention Comparison diagram;
Fig. 5 is the obtained voice quality sense using method Yu existing non-linear transform algorithm of the invention in example 1 Know the comparison diagram of assessment.
Specific embodiment
As shown in Figure 1, in hands-free communication system, the acoustical signal of distal end speaker by distal end stereo microphone It receives, obtains the stereo reference signal x in two channels1(n) and x2(n).Stereo reference signal x1(n) and x2(n) through past phase Pretreatment is closed, the signal x that obtains that treated1' (n) and x2'(n).Then pass through line transmission to proximal end, x1' (n) and x2'(n) It is played again by the stereophonic loud-speaker system of proximal end.The microphone channel signal of proximal end two is respectively d1(n) and d2(n), i.e. proximal end The signal that microphone receives, is divided into two parts: a part is x1' (n) and x2' after (n) played back by Near end speaker, Acoustical signal is propagated in proximal void, after reflection and scattering in near-end room, reaches the signal of proximal end microphone;Separately A part is ambient noise and voice in the scene of proximal end, i.e. near end signal.Proximal end microphone signal d1(n) and d2(n) in First part's signal can return to distal end by line transmission again, and be reset in the scene of distal end by far-end loudspeaker.So, Speaker in the scene of distal end has just heard the sound of a part oneself, and here it is the acoustic echos that needs are eliminated.
If signal vector corresponding to two channel speakers signals is respectively as follows:
x1' (n)=[x1′(n-L+1) x1′(n-L+2) ... x1′(n)]
x2' (n)=[x2′(n-L+1) x2′(n-L+2) ... x2′(n)]
The acoustic echo showed in the scene of distal end mainly due to caused by the acoustic propagation in the scene of proximal end, It is recognized in the scene of proximal end using adaptive channel and carries out acoustic echo elimination.h12(n) and h22(n) proximal end scene is respectively indicated In two loudspeaker channels to left microphone room impulse response,WithIt is then adaptive channel identification respectively Algorithm is to h12(n) and h22(n) estimation;h21(n) and h11(n) two loudspeaker channels are respectively indicated in the scene of proximal end to right biography The room impulse response of sound device,WithIt is then adaptive channel identification algorithm respectively to h21(n) and h11(n) estimate Meter.
Adaptive algorithm passes through
Obtain the residue signal e after echo cancellor1(n) and e2(n);e1(n) and e2(n) by vertical in the scene of distal end Body sound loudspeaker plays, and can so eliminate the acoustic echo occurred in the scene of distal end.
Filter imbalance is defined as:
Wherein, | | | |2Indicate 2 norms.
If pre-processing without decorrelation, biggish filter imbalance value η (n) will be present.That is, adaptive channel is distinguished Knowing algorithm, there are biggish filter evaluated errors.Especially when physical pulse response changes, big filter is lost Tune will lead to adaptive tracing and slow or even can not accurately track its variation, to make echo cancellor effect severe exacerbation. The reason is that if being pre-processed without decorrelation,
Due to stereo reference signal x1(n) and x2(n) the same speaker in the scene of distal end, between them With very strong Inter-channel Correlation.This strong correlation will make the convergency value of adaptive channel identification algorithm not unique, real The sef-adapting filter that border converges to when applyingIt may and be not equal to our desired physics Transmission function hij(n), i=1,2;J=1,2, it lacks of proper care so as to cause big filter.
In order to solve this problem, improve imbalance performance, usually in reference signal x1(n) and x2(n) it is transmitted to proximal end scene Decorrelation pretreatment is carried out to it before, so that filter imbalance be effectively reduced, improves stereo acoustic echo and eliminates performance.
The present invention will be further described in detail in the following with reference to the drawings and specific embodiments.
As shown in Fig. 2, a kind of interchannel decorrelation method in the elimination of stereo acoustic echo, specific steps are such as Under:
Step 1) distal end microphone pick-up of acoustic signals obtains stereo reference signal x1(n) and x2(n);Simultaneously to x1(n) And x2(n) Fourier transformation (FFT) is carried out, obtains frequency spectrum X1(k) and X2(k);
Step 2) is to frequency spectrum X1(k) and X2(k) high frequency decorrelative transformation is carried out;The time domain reference signal x that obtains that treated1d (n) and x2d(n);It specifically includes:
Simultaneously to frequency spectrum X1(k) and X2(k) high frequency decorrelative transformation is carried out:
Wherein, β (k) is nonlinear weight coefficient, it is contemplated that the relationship of the phase sensitive following frequency of human auditory system, β (k) value are as follows:
Wherein, f (k) is linear frequency value corresponding to discrete frequency indices k,NFFT is in Fu The length of leaf transformation;fsFor the sample frequency of signal;β (k) is in fsIt is maximized at/2;
Judge that analog frequency f (k) corresponding to k is greater than f0It is whether true, the f0It is high frequency or low frequency for division signals Critical value;Preferably, f0=1kHz;If a determination be made that negative, 0 is set by Θ value, otherwise, is set Θ value to The value of sinusoidal variations;
To frequency spectrum X1d(k) and X2d(k) carry out inverse Fourier transform (IFFT), obtain after high frequency decorrelative transformation when Domain reference signal x1d(n) and x2d(n)。
Step 3) is to three-dimensional acoustic reference signal x1(n) low frequency decorrelative transformation is carried out, three sinusoidal signals are constructed;Specifically Include:
Step 3-1) calculate stereo reference signal x1(n) weighted auto-correlation function ψw(τ);
x1(n) average magnitude difference function (AMDF) φ (τ) is defined as:
φ (τ)=E | x1(n)-x1(n+τ)|} (7)
Wherein, τ is time delay, and desired operation is sought in E { } expression;
x1(n) auto-correlation function is calculated as follows:
ψ (τ)=E { x1(n)x1(n+τ)}. (8)
Weighted auto-correlation function is are as follows:
ψw(τ)=ψ (τ)/[φ (τ)+ε], (9)
Wherein ε is the fixed value for preventing quotient from dissipating and introducing, preferred value ε=1.
Step 3-2) calculate fundamental frequency cycles estimated value τ0
Weighted auto-correlation function ψwThe maximum value ψ of (τ)w,maxCorresponding time delay number τ0The as estimated value of fundamental frequency cycles; The value range of fundamental frequency is limited to [60,350] Hz, this, which is equivalent to, limits τ0Search range, it may be assumed that
τ0={ τ: ψw(τ)=max (ψw(τ));fs/350≤τ≤fs/60} (10)
Wherein, max () expression is maximized.
Step 3-3) recalculate fundamental frequency estimated value f0
In order to improve the detection accuracy of fundamental frequency, further to maximum value ψw0) centered on adjacent 3 ψw(τ) value into Then row interpolation searches again for maximum value and obtains the higher fundamental frequency cycles estimated value T of precision0, then fundamental frequency value are as follows:
f0=1/T0 (11)
Step 3-4) retrieval reference signal frequency spectrum X1(k), it carries out amplitude weighting and obtains the amplitude A of three low order harmonics ingredientsi And first phase
k0Indicate frequency indices corresponding to fundamental frequency, then two, harmonics corresponds respectively to 2k three times0And 3k0;Utilize three frequencies Rate index value retrieves X1(k), the amplitude and first phase of fundamental frequency and two, harmonics three times are respectively obtained, amplitude weighting is carried out and obtains:
Wherein, A1And σ1For fundamental frequency amplitude, first phase and the amplitude weighting factor, A2And σ2For secondary harmonics amplitude, First phase and the amplitude weighting factor, A3And σ3For for the amplitude of harmonics three times, first phase and the amplitude weighting factor;Angle { } table Show and phase function, σ are taken to plural numberi(i=1,2,3) is in [0,1] interior value, it is preferred that σ1=0.1, σ2=0.2, σ3=0.3.
Step 3-5) three sinusoidal signal s of construction1(n)、s2(n) and s3(n):
Wherein, fi=if0, i=1,2,3.
Step 4) passes through time domain addition for three sinusoidal signal s1(n)、s2(n) and s3(n) it is added to x1d(n) in;? To x '1(n);x′2It (n) is x2d(n);x′1(n) and x '2(n) it is the stereo signal after Whole frequency band decorrelative transformation:
The signal x ' that step 5) obtains step 4)1(n) and x '2(n) it is carried out by normalized least fibre method adaptive Channel identification, realizes echo cancellor;
Within n-th of sampling period, the sef-adapting filter in n-th of sampling period is utilizedTo x '1(n) and x '2 (n) it filters, obtains the echo signal y at microphonej(n), it may be assumed that
Again in microphone signal dj(n) echo signal is eliminated in, obtains residue signal ej(n):
ej(n)=dj(n)-yj(n) (16)
Then sef-adapting filter is updated, obtains the sef-adapting filter in (n+1)th sampling period
Wherein, xi' (n)=[xi′(n-L+1) xi′(n-L+2) ... xi′(n)]。
For next sampling period, then continue to carry out adaptive tracing from formula (17) to formula (19), until signal terminates.
The normalized least fibre method (NLMS) can be other adaptive algorithms: least fibre method (LMS) or frequency Domain least fibre method (FBLMS).
Below with reference to example, the present invention will be further described.
Example 1: near-end room is having a size of 4 × 3 × 3m3, two Near end speakers be located at { 1,2,1.2 } m and 3,2, 1.2}m;The proximal end microphone being located at { 1.8,1,1.2 } m is investigated, the acoustical signal that it is picked up regards proximal end microphone letter as Number d (n).Signal-to-noise ratio is used to simulate near end signal for the white Gaussian noise of 30dB.Sample rate takes 16kHz, using NLMS algorithm into The identification of row adaptive channel.Method of the invention and traditional nonlinear transformation decorrelation method (NLT) are subjected to performance comparison.
Interchannel coherence factor (the Inter-Channel Coherence of two channel signal x1 (n) and x2 (n) Coefficient, ICCC) is defined as:
ICCC is in [0,1] interior value, and value is smaller, and inter-channel coherence is weaker, and decorrelation effect is better.As shown in figure 4, In the processing result of three methods, the ICCC value of treated by means of the present invention two signals is minimum.
In learning curve of lacking of proper care, it is shorter to reach stable state time-consuming, indicates that convergence rate is faster;Its value is smaller, and identification precision is got over It is high.If Fig. 4 shows, in the processing result of three methods, treated by means of the present invention misregistration signal best performance.
Perceptual evaluation of speech quality (Perceptual Evaluation of Speech Quality, PESQ) is to a variety of Human hearing characteristic modeling, by subjective sound quality objective quantification.PESQ value between -0.5 to 4.5, value is bigger, indicates distortion It is smaller.As shown in figure 5, in the processing result of three methods, the PESQ value ratio of treated by means of the present invention signal The PESQ value of signal is big after NLT processing.

Claims (5)

1. a kind of interchannel decorrelation method for stereo acoustic echo canceler, which comprises
Step 1) distal end microphone pick-up of acoustic signals obtains stereo reference signal x1(n) and x2(n);Simultaneously to x1(n) and x2 (n) Fourier transformation is carried out, frequency spectrum X is obtained1(k) and X2(k);
Step 2) is to frequency spectrum X1(k) and X2(k) high frequency decorrelative transformation is carried out;The time domain reference signal x that obtains that treated1d(n) and x2d(n);
Step 3) is to three-dimensional acoustic reference signal x1(n) low frequency decorrelative transformation is carried out, three sinusoidal signals: s are constructed1(n)、s2 (n) and s3(n);
Step 4) passes through time domain addition for three sinusoidal signal s1(n)、s2(n) and s3(n) it is added to x1d(n) in;It obtains x′1(n);x′2It (n) is x2d(n);x′1(n) and x '2It (n) is the stereo signal after Whole frequency band decorrelative transformation;
The signal x ' that step 5) obtains step 4)1(n) and x '2(n) adaptive channel identification is carried out by adaptive algorithm, it is real Existing echo cancellor.
2. the interchannel decorrelation method according to claim 1 for stereo acoustic echo canceler, which is characterized in that The step 2) specifically:
Simultaneously to frequency spectrum X1(k) and X2(k) high frequency decorrelative transformation is carried out:
Wherein, β (k) is nonlinear weight coefficient, it is contemplated that the relationship of the phase sensitive following frequency of human auditory system, β (k) Value are as follows:
Wherein, f (k) is linear frequency value corresponding to discrete frequency indices k,NFFT is Fourier transformation Length;fsFor the sample frequency of signal;β (k) is in fsIt is maximized at/2;
Judge that analog frequency f (k) corresponding to k is greater than f0It is whether true, the f0It is facing for high frequency or low frequency for division signals Dividing value;If a determination be made that negative, 0 is set by Θ value, otherwise, sets Θ value to the value of sinusoidal variations;
To frequency spectrum X1d(k) and X2d(k) inverse Fourier transform is carried out, the time domain reference signal after high frequency decorrelative transformation is obtained x1d(n) and x2d(n)。
3. the interchannel decorrelation method according to claim 2 for stereo acoustic echo canceler, which is characterized in that The f0Value are as follows: f0=1kHz.
4. the interchannel decorrelation method according to claim 2 for stereo acoustic echo canceler, which is characterized in that The step 3) specifically includes:
Step 3-1) calculate reference signal x1(n) weighted auto-correlation function ψw(τ);
x1(n) average magnitude difference function φ (τ) is defined as:
φ (τ)=E | x1(n)-x1(n+τ)|} (7)
Wherein, τ is time delay, and desired operation is sought in E { } expression;
x1(n) auto-correlation function is calculated as follows:
ψ (τ)=E { x1(n)x1(n+τ)}. (8)
Weighted auto-correlation function are as follows:
ψw(τ)=ψ (τ)/[φ (τ)+ε], (9)
Wherein ε is the fixed value for preventing quotient from dissipating and introducing, ε=1;
Step 3-2) calculate fundamental frequency cycles estimated value τ0
τ0={ τ: ψw(τ)=max (ψw(τ));fs/350≤τ≤fs/60} (10)
Wherein, max () expression is maximized;
Step 3-3) recalculate fundamental frequency estimated value f0
To with maximum value ψw0) centered on adjacent 3 ψw(τ) value carries out interpolation, then searches again for maximum value and obtains precision more High fundamental frequency cycles estimated value T0, then fundamental frequency value are as follows:
f0=1/T0 (11)
Step 3-4) retrieval reference signal frequency spectrum X1(k), it carries out amplitude weighting and obtains the amplitude A of three low order harmonics ingredientsiWith it is first Phase
k0Indicate frequency indices corresponding to fundamental frequency, then two, harmonics corresponds respectively to 2k three times0And 3k0, utilize three frequency ropes Draw value retrieval X1(k), the amplitude and first phase of fundamental frequency and two, harmonics three times are respectively obtained, amplitude weighting is carried out and obtains:
Wherein, A1And σ1For fundamental frequency amplitude, first phase and the amplitude weighting factor, A2And σ2Amplitude, first phase for secondary harmonics With the amplitude weighting factor, A3And σ3For for the amplitude of harmonics three times, first phase and the amplitude weighting factor;Angle { } expression pair Plural number takes phase function, σi, i=1,2,3 in [0,1] interior value, σ1=0.1, σ2=0.2, σ3=0.3;
Step 3-5) three sinusoidal signal s of construction1(n)、s2(n) and s3(n):
Wherein, fi=if0, i=1,2,3.
5. the interchannel decorrelation method according to claim 4 for stereo acoustic echo canceler, which is characterized in that The adaptive algorithm are as follows: least fibre method, normalized least fibre method or frequency domain least fibre method.
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