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1.
Although the signal subspace approach has been studied extensively for speech enhancement,no good solution has been found to identify signal subspace dimension in multichannel situation.This paper presents a signal subspace dimension estimator based on F-norm of correlation matrix,with which subspace-based multi-channel speech enhancement is robust to adverse acoustic environments such as room reverberation and low input signal to noise ratio (SNR).Experiments demonstrate the presented method leads to more noise reduction and less speech distortion comparing with traditional methods.  相似文献   

2.
A speech signal processing and features extracting method based on computational auditory model is proposed. The computational model is based on psychological, physiological knowledge and digital signal processing methods. In each stage of a hearing perception system, there is a corresponding computational model to simulate its function. Based on this model, speech features are extracted. In each stage, the features in different kinds of level are extracted. A further processing for primary auditory spectrum based on lateral inhibition is proposed to extract much more robust speech features. All these features can be regarded as the internal representations of speech stimulation in hearing system. The robust speech recognition experiments are conducted to test the robustness of the features. Results show that the representations based on the proposed computational auditory model are robust representations for speech signals.  相似文献   

3.
Microphone array-based speech enhancement has great importance for speech communications and speech recognition. To reduce the aperture of the microphone array and to increase the effect of the speech enhancement will greatly broaden the application areas of the microphone array. An array crosstalk resistant adaptive noise cancellation method is therefore presented. And then an improved spectral subtraction algorithm is further cascaded to obtain better enhancement results. Theoretic analysis and experiments indicate that the proposed scheme needs only a very small microphone array while it simultaneously achieves a higher SNR improvement. Besides, the proposed scheme can be used in many noisy environments and is easy for real-time implementation.  相似文献   

4.
周意  李澄 《中国物理 C》2008,32(Z2):232-234
Based on the principles of transmission line and the output signal of GEM detector, a full simulation model of delay-line circuit has been described in this paper. The consistency of simulation results and experimental data shows that the method is very effective for the design of delay-line readout.  相似文献   

5.
In this paper, a heuristic approach based on Slavic's peak searching method has been employed to estimate the width of peak regions for background removing. Synthetic and experimental data are used to test this method. With the estimated peak regions using the proposed method in the whole spectrum, we find it is simple and effective enough to be used together with the Statistics-sensitive Nonlinear Iterative Peak-Clipping method.  相似文献   

6.
Whispered speech enhancement using auditory masking model in modified Meldomain and Speech Absence Probability(SAP)was proposed.In light of the phonation characteristic of whisper,we modify the Mel-frequency Scaling model.Whispered speech is filtered by the proposed model.Meanwhile,the value of masking threshold for each frequency band is dynamically determined by speech absence probability.Then whispered speech enhancement is conducted by adaptively rectifying the spectrum subtraction coefficients using different masking threshold values.Results of objective and subjective tests on the enhanced whispered signal show that compared with other methods;the proposed method can enhance whispered signal with better subjective auditory quality and less distortion by reducing the music noise and background noise under the masking threshold value.  相似文献   

7.
It is well known that auditory system of human beings has excellent performance which automatic speech recognition(ASR) systems can’t match,and fractional Fourier transform (FrFT) has unique advantages in non-stationary signal processing.In this paper,the Gammatone filterbank is applied to speech signals for front-end temporal filtering,and then acoustic features of the output subband signals are extracted based on fractional Fourier transform. Considering the critical effect of transform order for FrFT,an order adaptation method based on the instantaneous frequency is proposed,and its performance is compared with the method based on ambiguity function.ASR experiments are conducted on clean and noisy Putonghua digits,and the results show that the proposed features achieve significantly higher recognition rate than the MFCC baseline,and the order adaptation method based on instantaneous frequency has much lower complexity than that based on ambiguity function.Further more,the FrFT-based features achieve the highest recognition rate using the proposed order adaptation method.  相似文献   

8.
In the past we analyzde the dynamic spectrum of infrasonic waves with FFT,but that isnot suitable to this analysis in any case.In this paper another method is proposed to analyzeinfrasonic waves.This method is that,after digital filtering of the signal and Hilbert transform,this paper the computation of the envelope with Hilbert transform and 1/12-octave filter are dis-cussed.Then a synthetic infrasonic signal generated by a large nuclear explosion is analyzed withthe two methods,and their results are compared.It is shown that,in the dynamic spectrum,usingHilbert transform,not only higher resolution is obtained,the main frequency range of separatedmodes can be achieved,but also their waveform can be obtained preliminarily for some modes.Thus it seems that in some cases this method is more efficient for discovering the characteristicsof an infrasonic signal.  相似文献   

9.
10.
Doppler effect widely exists in the signal from the moving acoustic source. In order to solve such problems as frequency shift and frequency band expansion, a time domain cor- rection method is presented in this paper. First, the discrete time vector for interpolation and the amplitude restoration formula is derived based on the moving relationship and the Morse acoustic theory, then the amplitude weights are corrected and the distortion signal is interpolated. Every point of the discrete signal is operated separately in time domain. Compared with the existing frequency domain methods, this method does not need to know the characteristic frequency beforehand and would not be influenced by the blending of the frequency band. Hence, this method can be employed to correct multiple frequency signals and it is also a simple and effective Doppler effect reduction method.  相似文献   

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