An Improved Speech Feature Extraction Algorithm Using DWT
Xiang Wu, Feng Tian, Jingao Liu
Department of Electronic Science & Technology, East China Normal University, 200241, Email:
knightwx@gmail.com
Abstract
This paper proposes a novel approach for speech
feature extraction., the proposed noise robust feature
extraction algorithm is found to be robust to different
noise characteristics, SNRs, and under various
training-testing matched or unmatched con- ditions.
And the algorithm this paper presents exhibit a low
computational cost. We also give a practical weighting
technique here. The experiment compares traditional
MFCCs algorithm and our methods, results show
overall improvement with the appli- cation of proposed
wavele based denoising algorithm with a modified
thresholding procedure.
1. Introduction
In pattern recognition, speech recognition is a very
important aspect. Particularly, with the com- puter
widely used, it appears to be even more significant, like
voice driven service portals, speech interfaces in
automotive navigational and guidance systems or
speech driven applications in modern offices [1].
MFCCs (Mel-Frequency Cepstral Coe- fficients) [2]
are important featrure parameter to the speech
recognition, but the performance of a speech
recognition system drops dramatically when there is a
mismatch between training and testing conditions.
Many different algorithms have been studied to
decrease the effect of noise on the recognition
performance [3]. [4, 5, 6] have given several useful
speech enhancement techniques.
Wavelet analysis is a rapid development of new
application in the mathematics field. It has become a
popular tool in many research areas. WT decomposes
data into a sparse, multiscale representation. The
wavelet transform, with its flexible time-frequency
resolution is, therefore, an appropriate tool for the
analysis of signals consisting of short high-frequency
bursts, and long quasi-stationary segments [7, 8]. This
paper uses WT in speech feature extraction, trying to
resolve the primary problem of acoustical mismatch for
978-1-4244-1724-7/08/$25.00 ©2008 IEEE
the degradation of automatic speech recognition
performance. Because of multi resolution we can have
a way of the lowest possible compu- tational costs in
realtime and good recognition per- formance.
The remainder of this paper is organized as follows.
In Section 2 wavelet transform in speech recognition is
briefly reviewed. In Section 3, we describe how the
features are extracted, different algorithm of all
processing steps and weighting are introduced in detail.
The experiment and comparison results of the orthodox
approach and our algorithm are given in Section 4.
Finally, we make a conclusion regarding our approach
and suggest some areas of further research in Section 5.
2. WT processing for speech signal
Feature extraction should execute after voice
activity detection. Amplifying, AD and denoising are
needed. VAD based on higher order statistics is
presented in [9]. The discrete time implementation of
wavelet transform is based on the iteration of
two-channel filter bank, which is on each level,
followed by a decimation-by-two unit [10]. While the
basis vectors of the DCT have approximately the same
resolution in time and frequency since same length
windows are used in calculating the cepstral
coefficients. Especially, DWT is suitable for noise
robustness issues. Wavelet coefficient sequence
thresholding combines simplicity and efficiency and,
therefore, it is an extensively investigated noise
reduction method. Fig.1 illustrates the block diagram of
the proposed algorithm.
Fig.1. Block diagram of proposed noise robust
feature extraction algorithm based on DWT
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 The DWT algorithm is given by Mallat [11]. Given In fact, Vaseghi has suggested multi-resolution
speech signal function f(x), wavelet base can be built features to overcome DCT problem in MFCCs ope-
as: ration in [12]. According to human hearing characte-
φ jk ( x) = 2 − j / 2 φ (2 − j x − k ) ristic as mel-frequency, the DWT should base on
(1) mel-frequency scaled bands. The parameter matrix is
− j/2 −j obtained by applying the DWT to the mel-scaled
ψ jk ( x) = 2 ψ (2 x − k)
(2) log-filterbank energies of a speech frame. The WT uses
{φ jk ( x)} {ψ jk ( x)} short basis functions to measure the high frequency
Where and are respectively content of the signal and long basis functions to
orthogonal function set. If P0f=f，in the level j Pj-1f can measure the low frequency content of the signal.
be decomposed using one-dimension DWT by
orthogonal projection Pjf and Qjf as： 3. Feature extracting and matching
Pj −1 f = Pj f + Q j f = ∑ c φ jk + ∑ d ψ k
j
k
j
jk
k k The following requirements to feature pa- rameters
(3) needs to be considered with the aim of improving
where： speech recognition performance:
p −1 p −1 1) The parameters can effectively present speech
c kj = ∑ h ( n ) c 2j k−1+ n d kj = ∑ g ( n ) c 2j k− +1 n feature.
n=0 , n=0 2) It is independent in each step.
( j = 1,2,..., L, k = 0,1,..., N 2 j − 1) 3) Low computational cost is needed for feature
(4) extracting and matching, effective feature post-
Where {h(n)} and {g(n)} are respectively low pass and processing algorithms should be applied to trans-
high pass weight coefficients. They are decided by form feature vectors in lower dimensional space,
{φ jk ( x)} {ψ jk ( x)} to decorrelate elements in feature vectors, and to
wavelet base and , p denotes enhance the classification process.
0
{C }n Mel-frequency wrapping use the following
the length of coefficients. are input data, N
approximate formula to compute the mels for a given
denotes signal length, L denotes the level. Hence 1D
frequency f in Hz:
DWT in each level is similar to 1D convolution except
sample distance. ⎛ f ⎞
f mel = 2595 log 10 ⎜ 1 + ⎟ (5)
After the operation, a 2D matrix about j and k is ⎝ 700 ⎠
gotten. Where j=1 denote approximate dissolution and
But MFCCs [13] derived on the basis of short-time
the speech spectrum from high to low was presented by
fourier transform and power spectrum estimation, have
j from low to high. K denotes the po- sition in time
been used predominantly to date as the funda- mental
domain. For k is different in different level, the feature
speech features in state-of-the-art speech recognition
parameter has pyramid structure, this will have less
systems. The STFT-based approach has, due to the
computational cost. The spreads of several level of ‘l’
constant analysis window length, fixed time-frequency
are represented in Fig. 2.
resolution and is, therefore, not optimized to analyze
simultaneously the non- stationary parts of a speech
signal with the same precision. Conventional algorithm
uses many filters in frequency domain to extract each
spectrum feature. They simply divided the frequency
band into subbands, extracted features for each
subband. So speech signal will be divided into short
time blocks and be transformed for each. Computation
in both time and frequency domain are referred, WT is
obviously more proper. The localization property of
transform can be explained by the feature parameters
matrix, it is

Fig.2. Speech signal and each level wavelet

decomposition

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 ⎡d 0 d 0 " d 0 " d 0 ⎤ 4. Experiment
⎢ 0 1 N /2 N
⎥
⎢ d 0 d1 " d N / 2
1 1 1
⎥ In this section, we describe the recognition
⎢ ⎥ experiments and their performances. Connected digit
⎢# % ⎥ (6) recognition experiments were performed using the
⎢d L " d L L ⎥ Aurora 2 [17] and Aurora 3 [18] databases, which were
⎢ 1 N / 2 −1 ⎥ designed to evaluate the performance of automatic
⎢c1L " c L L ⎥ speech algorithms under different noisy and acoustical
⎣ N / 2 −1 ⎦ mismatch conditions. The training and test files include
Conventional feature extraction methods use the 10 digits spoken 10 times each with approximately 1s,
entire frequency band to extract speech features for Background noises include speech and car. The same
speech recognition. The human speech recognition test and training files were used for all experiments.
system seems to utilize partial recognition infor- The speech signal was sampled at 8 kHz and
mation across frequencies, it’s local in frequency. analyzed with 24 ms hamming windows stepped by 8
According to the studies, voiced phonemes have the ms. For the computation of mel-scaled log filter-bank
maximum energy mainly at lower frequencies of the energies, 24 triangular mel-scaled band-pass filters
speech spectrum. Therefore, it was sufficient to were designed and implemented. With the method we
reconstruct the signal from the denoised subbands in proposed the log mel-filter bank outputs were replaced
the range 0-2 kHz only [14]. Here we design the WT with wavelet packet outputs (level 5). The feature
decomposition mode depends on human’s inner ear, parameters comparison is shown in Fig. 4.
which determines how much energy is contained at the
different frequencies that make up a specific sound
scene and when these energies occur in time [15]. As (5)
shows, the usual processing in MFCC calculation is to
use the Log compression of the mel filterbank outputs
as well as the Log compression of full frame energy.
We weighting the contribution of coefficients for the
total score based on Log function too. The weighting
coefficients are shown in Fig.3

Fig.3. Log compression weighting coefficients

Parameters matching is the same as con- ventional

algorithm, we use DTW (Dynamic Time Warping) [16] Fig.4. feature parameters Comparison
here. For noise affects each coefficient differently,
when one frequency band is corrupted by noise, only a The BER versus the SNR in different noise
few coefficients matching will be affected if the speech background of MFCCs and our feature parameters is
features represent local infor- mation as in our methods. shown in Fig. 5.
Even if the entire frequency band is corrupted, the
noise level can be different for each frequency interval
for real applications, so the various coefficients will be
affected differently. This significantly improves the
performance comparing to STFT.

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Fig. 5. BER performance of Speech recognition
by different feature parameters
The significant performance differences between
feature parameters experimentally demonstrates that
the use of local features in the frequency domain
improves the performance for the noisy speech. The
weighting algorithm significantly improve the per-
formance. These results suggest that weighting the
contribution of each coefficient to the recognition score
based on the noise level improves the per- formance.
5. Conclusion and further research
A new method for noise robust speech pa-
rameterization procedure based on DWT has been
presented. This algorithm can give compact and
accurate representation of speech signals in the time-
frequency plane, it also has efficient feature vector
post-processing. Compare to conventional MFCCs,
denoising algorithm using time-frequency adaptive
threshold estimation and modified soft thresholding
procedure, as well as the proposed voice activity
detection algorithm, is more efficient under high
discriminant speech-background noise conditions. At
last from the experiment result, it has been concluded
that the proposed denoising algorithm introduces lower
level of additional signal distortion due to proposed
time-frequency noise level adaptation and is, therefore,
more appropriate for the effective noise reduction of
noisy speech signals.
6. Acknowledgment
This work was supported in part by the key project
in Shanghai Science and Technology Committee under
1089
Grant 075115002. Authors wish to thanks Pr. Liu
Jin-gao whose comments and su- ggestions have
largely contributed to improve this paper.
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