CN101916567A - Speech enhancement method applied to dual-microphone system - Google Patents
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Abstract
The invention provides a speech enhancement method applied to a dual-microphone system. The dual-microphone system comprises a microphone array module, a control module, a processing module and an output module, wherein the microphone array module consists of a first microphone and a second microphone; the control module is used for receiving a signal transmitted by the microphone array module and controlling a microphone array; the processing module is used for receiving data transmitted by the control module; and the output module is used for outputting the processed data output by the processing module. The speech enhancement method comprises the following step that: 1, the first microphone and the second microphone receive time domain speech signals with noise (S1) and (S2) and transmit the signals to the control module, and the control module performs framing and pre-emphasis processing on the time domain speech signals with noise (S1) and (S2) and converts the time domain speech signals with noise (S1) and (S2) into frequency domain signals with noise (X1) and (X2) through short-time Fourier transform. Therefore, the attenuation on related noise is improved.
Description
[technical field]
The present invention relates to a kind of sound enhancement method, relate in particular to the method that a kind of voice that are applied to dual-microphone system strengthen.
[background technology]
Because the existence of a large amount of neighbourhood noises, the general signal to noise ratio (S/N ratio) of the voice signal that the microphone of communication apparatus such as mobile phone collects is not high enough, particularly in high-noise environments such as street automobile, needs to improve volume the other side is not heard.So need promote the signal to noise ratio (S/N ratio) of input voice by the method that voice strengthen, improve communication quality.
Yet traditional single channel sound enhancement method noise reduction is limited, and can cause big distortion to voice.Use two-microphone array, can introduce the phase information of voice and noise signal, the effect of raising squelch also reduces the distortion of voice.
[summary of the invention]
The technical problem to be solved in the present invention provides a kind of sound enhancement method.
For solving the problems of the technologies described above, technical scheme provided by the invention is:
A kind of sound enhancement method that is applied to dual-microphone system, this dual-microphone system comprises
The microphone array module that constitutes by first microphone and second microphone;
Be used to receive signal that this microphone array module sends and the control module of controlling this microphone array;
Be used to receive the processing module of the data that control module sends;
Be used for output module that the data of processing module output are exported after treatment;
This method comprises the steps:
Step 1. first microphone and second microphone receive time domain Noisy Speech Signal (S1) respectively, (S2) after, send control module to, by control module to time domain Noisy Speech Signal (S1), (S2) carry out respectively branch frame, pre-emphasis handle, through Short Time Fourier Transform with time domain Noisy Speech Signal (S1), (S2) be not transformed into frequency domain signal with noise (X1), (X2); Make when wherein dividing frame between the time domain signals with noise of adjacent two frames the aliasing part is arranged;
By the priori snr value of processing module, obtain present frame frequency domain Noisy Speech Signal (X1) or decay (X2) gain according to the present frame that obtains; Gain with the above-mentioned decay that obtains by processing module, multiply by the auto-power spectrum of first microphone or the second microphone frequency domain signal with noise, the auto-power spectrum of the clean speech estimated signal after obtaining handling;
Decay gain by former frame obtains present frame frequency domain signal with noise (X1), noise cross-power frequency spectrum (X2);
Obtain present frame frequency domain signal with noise (X1) by processing module by the noise cross-power spectrum that obtains, posteriority signal to noise ratio (S/N ratio) (X2), and obtain the priori snr value of present frame, export to output module;
Frequency-region signal after step 3. will be handled by output module transforms to time domain, and the processing of postemphasising becomes output signal.
Beneficial effect of the present invention is, use two-microphone array, calculate the cross-power spectrum that dual microphone receives signal, suppress uncorrelated noise, and by from the signal cross-power spectrum, deducting the noise cross-power spectrum, and the calculating of priori SNR estimation value, having improved decay for correlation noise, performance is better than existing single channel voice enhancement algorithm.
[description of drawings]
Fig. 1 is the structured flowchart of the system of sound enhancement method application of the present invention;
Fig. 2 is the schematic flow sheet of voice enhancement algorithm of the present invention.
[embodiment]
The present invention will be further described below in conjunction with accompanying drawing.
A kind of sound enhancement method that is applied to dual-microphone system, referring to Fig. 1, this dual-microphone system comprises,
The microphone array module that constitutes by first microphone and second microphone;
Be used to receive signal that this microphone array module sends and the control module of controlling this microphone array;
Be used to receive the processing module of the data that control module sends;
Be used for output module that the data of processing module output are exported after treatment;
Comprise the steps:
Step 1. first microphone and second microphone receive time domain Noisy Speech Signal S1 respectively, behind the S2, send control module to, by control module to time domain Noisy Speech Signal S1, S2 carry out respectively branch frame, pre-emphasis handle, through Short Time Fourier Transform with time domain Noisy Speech Signal S1, S2 is transformed into frequency domain signal with noise X1, X2 respectively; Make when wherein dividing frame between the time domain signals with noise of adjacent two frames the aliasing part is arranged;
By the priori snr value of processing module, obtain the decay gain of present frame frequency domain Noisy Speech Signal X1 or X2 according to the present frame that obtains; Gain with the above-mentioned decay that obtains by processing module, multiply by the auto-power spectrum of first microphone or the second microphone frequency domain signal with noise, the auto-power spectrum of the clean speech estimated signal after obtaining handling;
Decay gain by former frame obtains present frame frequency domain signal with noise X1, the noise cross-power frequency spectrum of X2;
Obtain present frame frequency domain signal with noise X1 by processing module by the noise cross-power spectrum that obtains, the posteriority signal to noise ratio (S/N ratio) of X2, and obtain the priori snr value of present frame, export to output module;
Frequency-region signal after step 3. will be handled by output module transforms to time domain, and the processing of postemphasising becomes output signal.
Concrete, referring to Fig. 2, the voice enhancement algorithm basic step among the present invention is as follows:
1. the Noisy Speech Signal that dual microphone is received carries out the branch frame respectively, and pre-emphasis is handled, and arrives frequency domain through Short Time Fourier Transform;
2. the respectively auto-power spectrum and the cross-power spectrum of two Noisy Speech Signals of computational transformation behind the frequency domain, and the noise cross-power spectrum that obtains estimating by the decay gain of former frame signal;
3. calculate the posteriority signal to noise ratio (S/N ratio) of current frame signal by signal and the noise cross-power spectrum that obtains previously, and obtain the priori SNR estimation value of present frame by the priori SNR estimation value of former frame;
4. according to the priori SNR estimation value that obtains, calculate the decay gain of current frame signal;
5. with the decay that obtains gain, one road microphone signal frequency spectrum is wherein handled;
6. the frequency-region signal after will handling transforms to time domain, and the processing of postemphasising becomes output signal.
In the case introduction of following mask body, the sampling rate of the noisy speech signal of speech-enhancement system input is 16kHZ, and resolution is 16.
To the time domain Noisy Speech Signal x1 that two microphones receive, x2 carries out the branch frame.Be meant with Noisy Speech Signal to be that unit is divided into some signals with noise unit with the frame.Described signals with noise unit is made up of sampled point, chosen the sample frequency of 16kHz among the present invention, needs according to the short-time spectrum analysis, frame length is generally set between 10～35ms, present embodiment is divided frame with 16ms, and promptly a frame signals with noise unit is provided with 256 sampled points, naturally, any frame signals with noise unit has certain frame length, and the frame length of arbitrary frame is 256 among the present invention.
For the blocking effect between the signals with noise unit that prevents adjacent two frames, when minute frame, to make between the signals with noise unit of adjacent two frames certain aliasing part is arranged, that is, it is former frame section data data that D data are arranged in these frame data, and wherein aliasing partly is described below:
s
i(n)＝d
i(m，D+n) 0≤n＜L，i＝1，2
S wherein
iExpression input tape noisy speech signal, i gets 1 and 2 and represents two paths of signals respectively
d
i(m，n)＝d
i(m-1，L+n) 0≤n＜D
Wherein, d
i256 point sampling signals of expression present frame, because the length of any frame is 256, Duplication is 75%, so the sampled point number D=192 of lap.Distance L=256-192=64 that first sampled point of the signals with noise unit of consecutive frame is separated by.
Can have 50%～75% Duplication between the signals with noise unit of adjacent two frames of the present invention.Present embodiment is chosen between the signals with noise unit of adjacent two frames has 75% Duplication, promptly consistent with the Noisy Speech Signal unit of 75% (192 point) after the signals with noise unit of preceding 75% (192 point) of this frame and the former frame.
Two paths of signals behind the branch frame passes through same Hi-pass filter respectively, handles as pre-emphasis.Because ground unrest is generally bigger at the low frequency part energy in the voice signal,, make the enhancing better effects if so use can the decay component of low frequency part of described Hi-pass filter.Its form is as follows:
H(z)＝1-αz
-1
The general value of α is between 0.75-0.95, and effect preferably can be obtained in α=0.9 here.
Because voice signal is stably in short-term, is feasible so signal is carried out the processing of branch frame, but divides frame can bring the discontinuous frequency that causes of frame signal boundary to reveal again.So will carry out Short Time Fourier Transform (STFT) here.Short Time Fourier Transform can be understood as does Fourier transform again to the windowing of frame signal elder generation.The purpose of windowed function is exactly for when doing Short Time Fourier Transform, reduces the discontinuous frequency that causes of frame signal boundary and reveals.Here used a length to equal the Hamming window of 256 of frame lengths, it can effectively reduce the oscillation degree of Gibbs' effect.
Hamming window function is defined as follows:
win(n)＝{
0.54-0.46cos(2*π*n/M) 0≤n≤M-1
0 all the other n
}
Then Short Time Fourier Transform is as follows
Wherein, M=256 is the computational length of Fourier Tranform in short-term.M represents the m frame signal.I=1,2 expression two paths of signals.
So just with the Noisy Speech Signal s of present frame
iTransform from the time domain to and be frequency domain signal X
i
Then use following formula to calculate the auto-power spectrum and the cross-power spectrum of two-way Noisy Speech Signal, consider the continuity between frame and the frame simultaneously, carry out level and smooth each signal energy spectrum:
Wherein, m represents the sequence number of present frame, and f represents through different Frequency point after the FFT conversion, λ
x=0.6 is smoothing factor.
P
XiXjThe energy spectrum of the signal of expression after smoothly.
Work as i=j=1, expression microphone 1 receives the auto-power spectrum of signal;
Work as i=j=2, expression microphone 2 receives the auto-power spectrum of signal;
As i=1, j=2, expression microphone 1 and microphone 2 receive the cross-power spectrum of Noisy Speech Signal.
Cross-power spectrum to the present frame noise signal is estimated
Wherein, λ
n=0.9 is smoothing factor, and (f m-1) represents the decay factor that former frame calculates to q.
Then, calculate the posteriority SNR estimation of current frame signal by the Noisy Speech Signal cross-power spectrum that obtains previously and noise signal cross-power spectrum, as follows
Based on the priori SNR estimation formula of Ephraim and Malah, calculate the priori SNR estimation value of present frame then
Next, the decay calculating of gain factor.Here the priori SNR estimation value that draws based on previous calculations is taked different strategies.
For the big frequency of signal to noise ratio (S/N ratio), can think voice signal, adopt the method for spectral substraction to obtain decay factor, for the little frequency of signal to noise ratio (S/N ratio), think noise signal, it is carried out to a certain degree decay.Select SNR among the present invention
Prior＞1.5 is criterion, is higher than 1.5 the voice signal of thinking, is lower than 1.5 and can thinks noise.
Concrete gain reduction factor computing formula is as follows,
Noisy Speech Signal X with microphone 1 present frame
1(f m), multiply by the decay gain factor of the corresponding frequencies point that obtains previously, and what obtain is exactly voice signal after the enhancing of this frequency.
Wherein,
Be the voice signal estimated value after f frequency enhancing of m frame.
The operation that last spatial transform and output carry out has:
The first step: inverse fast fourier transform (FFT) transforms to time domain to the speech manual of frequency domain, the time domain voice signal after being enhanced.
The conversion of time domain realizes with general contrary discrete Fourier transform (IDFT).
Wherein, M=256 is frame length.S is the voice signal that transforms to after full range band after the time domain strengthens.
Second step: the processing of postemphasising.
With the pre-emphasis of front handle opposite, here with signal by a low-pass filter, farthest reduce original signal.The frequency response of wave filter is as follows;
H(z)＝1+αz
-1
The coefficient here is corresponding with the processing of front pre-emphasis, gets α=0.9.
The 3rd step: the lap of the consecutive frame of the voice signal after will strengthening carries out the phase add operation.
Concrete lap addition can be represented with following method.
L=64 is the distance that adjacent frame signal begins to locate, and M=256 is frame length.The final output signal after the phase add operation is finished in s ' representative.
Compare with correlation technique, it is the voice of being correlated with and the hypothesis of uncorrelated noise signal that dual microphone voice enhancement algorithm that the present invention proposes has effectively utilized that dual microphone receives, suppressed the interference of uncorrelated noise in the voice signal, simultaneously by from the signal cross-power spectrum, deducting the noise cross-power spectrum, and the calculating of priori SNR estimation value, improved decay for correlation noise.
The above only is a better embodiment of the present invention; protection scope of the present invention is not exceeded with above-mentioned embodiment; as long as the equivalence that those of ordinary skills do according to disclosed content is modified or changed, all should include in the protection domain of putting down in writing in claims.
Claims (6)
1. sound enhancement method that is applied to dual-microphone system, this dual-microphone system comprises
The microphone array module that constitutes by first microphone and second microphone;
Be used to receive signal that this microphone array module sends and the control module of controlling this microphone array;
Be used to receive the processing module of the data that control module sends;
Be used for output module that the data of processing module output are exported after treatment;
It is characterized in that comprising the steps:
Step 1. first microphone and second microphone receive time domain Noisy Speech Signal (S1) respectively, (S2) after, send control module to, by control module to time domain Noisy Speech Signal (S1), (S2) carry out respectively branch frame, pre-emphasis handle, through Short Time Fourier Transform with time domain Noisy Speech Signal (S1), (S2) be not transformed into frequency domain signal with noise (X1), (X2); Make when wherein dividing frame between the time domain signals with noise of adjacent two frames the aliasing part is arranged;
Step 2. receives frequency domain Noisy Speech Signal (X1) by processing module, (X2), and obtains frequency domain Noisy Speech Signal (X1) respectively, (X2) auto-power spectrum and frequency domain Noisy Speech Signal (X1), and cross-power spectrum (X2),
By the priori snr value of processing module, obtain present frame frequency domain Noisy Speech Signal (X1) or decay (X2) gain according to the present frame that obtains; Gain with the above-mentioned decay that obtains by processing module, multiply by the auto-power spectrum of first microphone or the second microphone frequency domain signal with noise, the auto-power spectrum of the clean speech estimated signal after obtaining handling;
Decay gain by former frame obtains present frame frequency domain signal with noise (X1), noise cross-power frequency spectrum (X2);
Obtain present frame frequency domain signal with noise (X1) by processing module by the noise cross-power spectrum that obtains, posteriority signal to noise ratio (S/N ratio) (X2), and obtain the priori snr value of present frame, export to output module;
Frequency-region signal after step 3. will be handled by output module transforms to time domain, and the processing of postemphasising becomes output signal.
2. the sound enhancement method that is applied to dual-microphone system according to claim 1 is characterized in that: in the step 1, the frame length when dividing frame is set between 10～35ms; Described pre-emphasis is treated to the time domain Noisy Speech Signal (S1) through undue frame, the component of the low frequency part that (S2) decays respectively; Between the time domain signals with noise of adjacent two frames lap is arranged, its Duplication is 50%～75%.
3. the sound enhancement method that is applied to dual-microphone system according to claim 2, it is characterized in that: aliasing partly is described below:
s
i(n)＝d
i(m，D+n) 0≤n＜L，i＝1，2
S wherein
iExpression time domain Noisy Speech Signal,
d
i(m，n)＝d
i(m-1，L+n) 0≤n＜D
Wherein, d
iThe sampled signal of expression present frame, D represents the sampled point number of lap; L represents the distance that first sampled point of the time domain Noisy Speech Signal of consecutive frame is separated by, and m represents current frame number, and n represents a certain point data in the present frame.
4. the sound enhancement method that is applied to dual-microphone system according to claim 2 is characterized in that: time domain Noisy Speech Signal (S1), and (S2) decay the respectively branch metering method of low frequency part is:
H(z)＝1-αz
-1
Wherein, the general value of α is between 0.75-0.95, and H (z) represents transport function,
5. the sound enhancement method that is applied to dual-microphone system according to claim 1 is characterized in that:
In the step 1, the method for revealing for the discontinuous frequency that causes that reduces the frame signal boundary in the Short Time Fourier Transform process is: to the windowing of frame signal elder generation,
win(n)＝{
0.54-0.46cos(2*π*n/M) 0≤n≤M-1
0 all the other n
}
Win (n) represents employed Hamming window, and M represents in short-term that the computational length n of Fourier Tranform represents wherein a certain data point:
The method of Short Time Fourier Transform is:
Wherein, X
i(f, m) representation transformation is to the two paths of signals of frequency domain; M represents the m frame signal; The wherein a certain time domain data point of frame length n representative of Fourier transform is done in the M representative, and f represents a certain Frequency point of frequency domain; I gets 1 or 2.
6. the sound enhancement method that is applied to dual-microphone system according to claim 1 is characterized in that:
In the step 2, obtain frequency domain signal with noise X1, X2 auto-power spectrum and frequency domain signal with noise X1, the method for the cross-power spectrum of X2 is:
Wherein, m represents the sequence number of present frame, and f represents through different Frequency point behind the Fourier Tranform in short-term, λ
x=0.6 is smoothing factor;
P
XiXjThe energy spectrum of the signal of expression after smoothly; X
i(f, m) representation transformation is to the two paths of signals of frequency domain; M represents the m frame signal; X
j *(f, m) representation signal is got conjugation;
Work as i=j=1, represent the auto-power spectrum of the frequency domain signal with noise X1 that first microphone receives;
Work as i=j=2, represent the auto-power spectrum of the frequency domain signal with noise X2 signal that second microphone receives;
As i=1, j=2, represent that first microphone and second microphone receive frequency domain signal with noise X1, the cross-power spectrum of X2;
The method that obtains present frame noise cross-power frequency spectrum is
Wherein, λ
n=0.9 is smoothing factor, and (f m-1) represents the decay gain factor that former frame calculates and stores, P to q
N1N2(m represents the sequence number of present frame for f, the m) cross-power spectrum of expression present frame, and f represents different Frequency points;
The method that obtains the posteriority signal to noise ratio (S/N ratio) of current frame signal is:
SNR wherein
Post(m represents the sequence number of present frame for f, m) the posteriority signal to noise ratio (S/N ratio) of expression present frame, and f represents different Frequency points.
The method that obtains the priori signal to noise ratio (S/N ratio) of present frame is:
Wherein
The priori signal to noise ratio (S/N ratio) of expression present frame, m represents the sequence number of present frame, and f represents different Frequency points, and (f m-1) represents the decay gain factor that former frame calculates and stores, P to q
N1N2(f, the m) cross-power spectrum of expression present frame, SNR
Post(α is a smoothing factor for f, m) the posteriority signal to noise ratio (S/N ratio) of expression present frame, and general value is 0.7～0.9.
The decay gain obtains method and is:
Wherein q (f, m) expression decay gain factor,
The priori signal to noise ratio (S/N ratio) of expression present frame, m represents the sequence number of present frame, f represents different Frequency points, P
NiNj(f, m) (mutually) power spectrum certainly of expression present frame, SNR
Post(f, m) the posteriority signal to noise ratio (S/N ratio) of expression present frame.
In the step 2, (f m), multiply by the decay gain factor of the corresponding frequencies point that former frame obtains, and what obtain is exactly voice signal after the enhancing of this frequency with the Noisy Speech Signal X1 of the first microphone present frame
Wherein,
Be the voice signal estimated value after f frequency enhancing of m frame, q (f, m) expression decay gain factor.
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