ICASSP 2006 Robustness Techniques Survey ShihHsiang 2006

PARAMETRIC NONLINEAR FEATURE EQUALIZATION FOR ROBUST SPEECH RECOGNITION Luz Garc’ıa, Jos’e C. Segura, Javier Ram’ırez, Angel de la Torre, Carmen Ben’ıtez Dpto. Teor’ıa de la Se˜nal, Telem’atica y Comunicaciones (TSTC) Universidad de Granada

3 Introduction HEQ have been successfully applied to deal with the nonlinear effect of the acoustic environment in the feature domain –Normalize the probability distributions of the features in such a way that the acoustic environment effects are (partially) removed HEQ still suffer from several limitations –Rely on a local estimation of the probability distributions of the features based on a reduced number of observations belonging to a single utterance to be equalized –Nonlinear transformation is based on mapping the global CDF of each feature into a reference one –The transformations are usually based on a component-by- component equalization of the feature vector, thus discarding any cross-information between features in the equalization process

4 Introduction (cont.) In this paper, a parametric nonlinear equalization technique is proposed –Relies on a two Gaussian model for the probability distribution of the features –And on a simple Gaussian classifier to label the input frames as belonging to the speech or non-speech classes Recognition experiments on the AURORA 4 database have been performed and the effectiveness of the algorithm is analyzed in comparison with other linear and nonlinear feature equalization techniques

5 Review Histogram Equalization For a given random variable y with probability density function, a function mapping into a reference distribution

6 Review Histogram Equalization (cont.) The relative content of non-speech frames is a cause of variability in the HEQ transformation –because an estimation of the global probability distribution is used, that takes into account both speech and non-speech frames

7 Review Histogram Equalization (cont.) The unwanted variability of the transformation induced by the variable proportion of non-speech frames of each utterance can be –Reduced by removing non-speech frames before the estimation of the transformation –Another possibility is to use different transformations for speech and non-speech frames Instead of using a transformation to map the global CDF’s of the features, we can build separate mappings for speech and non-speech frames –As an alternative, the propose the use of a parametric form of the equalization transform based on a two Gaussian mixture model The first Gaussian is used to represent non-speech frames, while the second one represents speech frames

8 Two-class parametric equalization For each class, a parametric linear transformation is defined to map the clean and noisy representation spaces The clean Gaussians for speech and non-speech frames can be estimated from the training database, while the noisy Gaussians should be estimated from the utterance to be equalized non-speech speech clean Gaussian noise noisy Gaussian noisy speech

9 Two-class parametric equalization (cont.) In order to select whether the current frame y is speech or non-speech, a voice activity detector could be used –Implies a hard decision between both linear transformations that could create discontinuities in the limit of the non-speech/speech decision Instead, a soft decision can be used The posterior probabilities P(n|y) and P(s|y) are obtained using a simple two-class Gaussian classifier on MFCC C0

10 Two-class parametric equalization (cont.) Training the two-class Gaussian classifier –Initially, those frames with C0 below the mean value are assigned to the non-speech class and those with C0 above the mean are assigned to the speech class –The EM algorithm is then iterated until convergence (usually, 10 iterations are enough) to obtain the final classifier This classifier is used to obtain the class probabilities P(n|y) and P(s|y) and also to obtain the mean and covariance matricesμ n,y 、 Σ n,y 、 μ s,y and Σ s,y for the non- speech and speech classes for the given noisy input utterance

11 Two-class parametric equalization (cont.) The two Gaussian model for the C0 and C1 cepstral coefficients (used as reference model) along with the histograms of the speech and non-speech frames for a set of clean utterances

12 Experimental Results The proposed parametric equalization algorithm has been tested on the AURORA4 (WSJ0) database –The recognition system used in all cases is based on continuous crossword triphone models with 3 tied states and a mixture of 6 Gaussians per state –The language model is the standard bigram for the WSJ0 task –A feature vector of 13 cepstral coefficients is used as the basic parameterization of the speech signal using C0 instead of the logarithmic energy –The baseline reference system (BASE) uses sentence-by-sentence subtraction of the mean values of each cepstral coefficient (CMS) –The parameters of the reference distribution have been obtained by averaging over the whole clean training set of utterances

13 Experimental Results First row (BASE) corresponds to the baseline system which is based on a simple CMS linear normalization technique. The second row (HEQ) shows the word error rates when using a standard quantile-based implementation of HEQ –relative word error reduction of 17.8% The performance of HEQ is clearly improved by PEQ as shown in the third row, with a relative word error reduction of 30.8%. This result is very close to the one obtained for the AFE, which yields a 31.4% reduction of the word error rate Moreover, PEQ outperforms AFE in half of the tests (i.e. 02, 06, 08, 09, 10, 11 and 13).

14 Conclusions and Future Work The transformation is based on a nonlinear interpolation of two independent linear transformations –The linear transformations are obtained using a simple Gaussian model for the classes of speech and non-speech features The technique evaluated on a complex continuous speech recognition task showing its competitive performance against linear and nonlinear feature equalization techniques like CMS and HEQ A study of influence of within class cross-correlations is currently under development

MODEL-BASED WIENER FILTER FOR NOISE ROBUST SPEECH RECOGNITION Takayuki Arakawa, Masanori Tsujikawa and Ryosuke Isotani Media and Information Research Laboratories, NEC Corporation, Japan

16 Introduction Various kinds of background noise exist in the real world –Therefore robustness against various kinds of noise is quite important. Several approaches have been proposed to deal with this issue –Signal-processing-based spectral enhancement Spectral Subtraction (SS), Wiener Filter (WF) Less computational costs, but needs many tuning costs depending on the kind of noise and signal-to-noise ratio (SNR) –Statistical-model-based noise adaptation The acoustic model i.e., a hidden Markov model (HMM), is adapted to the noisy environment It needs huge computational costs to adapt the distributions to a noisy environment

17 Introduction (cont.) –Statistical-model-based compensation Using Gaussian mixture model (GMM) The computational cost is still much more than that of the signal-processing-based spectral enhancement. In this paper, they proposed Model-Based Wiener filter (MBW) Concept

18 Proposed Method (Cont.) A GMM with K Gaussian distributions is used as knowledge of clean speech in the cepstrum domain MBW algorithm

19 Proposed Method (Cont.) The noisy speech signal X(t) is modeled as Step 1: Perform Spectral Subtraction (SS) Step 2: Derive the expected value of the clean speech noisy speechclean speechnoise temporary clean speech estimated noise Spectrum Domain Cepstrum Domain MMSE Estimation flooring parameter

20 Proposed Method (Cont.) Step 3: Calculate Wiener Gain Step 4: Get the final estimated clean speech smoothing parameter

21 Experiments and Results Experiment Condition –The Mel-frequency cepstral coefficients (MFCC) and their 1st and 2nd derivatives are used as feature value of speech (include C0) –The feature value for GMM is composed of a 13-dimensional MFCC only –The flooring parameter α is set at 0.1, and the smoothing parameter β is set at 0.98 The MBW method was tested on the Aurora2-J task –contains utterances (in Japanese) of consecutive digit string recorded in clean environments –The other conditions are the same as Aurora2

22 Experiments and Results (cont.) The performance of different mixture number of the GMM 5dB restaurant noise At the point of 128 or 256, it becomes saturated

23 Experiments and Results (cont.) The word accuracy for each SNR almost equivalent to that of the AFE.

24 Experiments and Results (cont.) The Word Accuracy over the SNR for each kind of noise. These results show that the proposed method is much more robust than the AFE against various kinds of noise.

25 Conclusions Review MBW algorithm –Roughly estimates clean speech signals using SS –Compensates them using a GMM to improve robustness against non-stationary noise –The compensated speech signal is used to calculate the Wiener gain –Performing Wiener filtering The results show that the proposed method performs as well as the ETSI AFE These results demonstrate that the proposed method is robust against various kinds of noise