A Comparative Study between Discrete Wavelet Transform and Linear Predictive
Coding
D.Ambika
Research Scholar,
Department of Computer Science,
Avinashilingam Institute for Home Science and Higher
Education for Women, Coimbatore, India
ambikaphdscholar@gmail.com
Abstract— In this paper the analysis of the compression
process was performed by comparing the compressed signal
against the original signal. To do this the most powerful speech
analysis and compression techniques such as Linear Predictive
Coding (LPC) and Discrete Wavelet Transform (DWT) was
implemented using MATLAB. Here nine samples of spoken
words are collected from different speakers and are used for
implementation. The results obtained from LPC were
compared with other compression technique called Discrete
Wavelet Transform. Finally the results were evaluated in
terms of compressed ratio (CR), Peak signal-to-noise ratio
(PSNR) and Normalized root-mean square error
(NRMSE).The result shows that DWT performance was better
for these samples than the LPC method.
Keywords- Speech compression, LPC, CR,DWT, PSNR, NRMSE.
I. INTRODUCTION
Compression algorithm helps to reduce the bandwidth
requirement and also provide a level of security for the data
being transmitted. It is more important in teleconferencing
and wireless communication. Here it is more important to
ensure that compression schemes retain the integrity of the
speech. If the data is distorted in some way, it becomes
difficult to understand [1]. Thus, speech compression needs
to be performed in a way which retains the key qualities of
the data. Speech compression finds application in mobile
satellite communication, cellular phones and in audio
conferencing system etc.Today applications of speech coding
and compression have become very numerous. Compression
techniques can be classified into one of the two main
categories: Lossless and Lossy. In Lossless compression, the
original file can be perfectly recovered from the compressed
file [2]. In case of Lossy compression, the original file
cannot be perfectly recovered from the compressed file, but
it gives best possible quality for the compression than
lossless by discarding less-critical data. Speech coding is a
lossy type of coding, in which the output signal does not
exactly sound like the input signal.
The paper is organized as follows. Section 2 explains the
Types of speech coding techniques. Section 3 explains the
speech compression using LPC. Section 4 deals with speech
978-1-4673-4804-1 2012
c IEEE
V.Radha
Associate Professor,
Department of Computer Science,
Avinashilingam Institute for Home Science
and Higher Education for Women, Coimbatore, India
radhasrimail@gmail.com
compression based on DCT. Section 5 explores the
performance evaluation for the adopted techniques. Finally,
the conclusion is summarized in section 6.
II. TYPES OF SPEECH CODING TECHNIQUES
Although human beings have an audible frequency range
of 20Hz–20kHz, the human speech has significant frequency
components only up to 4 kHz, a property that is exploited in
the compression of speech [1]. The “fig 1” shows the various
methods for coding the speech signal [2].
Several techniques of speech coding such as LPC,
Waveform coding and Sub Band coding exist. Waveform
coding is used to analyze code and reconstruct original
speech sample by sample. It includes time domain coding
and frequency domain coding. The method such as Pulse
Code Modulation (PCM), Differential PCM (DPCM) [3],
Adaptive DPCM (ADPCM), Delta Modulation (DM), and
Adaptive PCMID are some of the popular time domain
waveform coding techniques and Transform Coding (IC),
Sub band Coding (SBC) are a few spectral domain waveform
coding techniques.
The PCM [3] is used to digitize the signals through signal
conversion. The DPCM can be analog signal or a digital
signal. It uses the baseline of PCM but it adds some
functionality based on the prediction of the samples of the
signal. In DPCM, first an estimate of each sample is found
based on prediction from past few samples and then the
difference of estimate made from the original. The DPCM
can provide PCM quality of speech at 56kbps. The Adaptive
Differential Pulse Code Modulation (ADPCM) [3] which is
used to provide much lower data rates by using a functional
model of the human speaking mechanism at the receiver end
[4]. The frequency domain includes sub band coding and the
transform coding. In transform coding the signal is
transformed to its representation in another domain in which
it can be compressed well than in its original form. This type
of coding uses the information about human vocal and
auditory systems. Using the transformation schemes such as
Discrete Cosine Transform (DCT) and Discrete Wavelet
Transform (DWT), the important frequency components can
be encoded with more precision than others.
966

Figure 1. Types of speech coding techniques
In this paper the two promising techniques such as LPC and
the transformation technique DWT are used for the coding of
speech signal and it is evaluated for its performance.
III. DISCRETE WAVELET TRANSFORM
For analyzing and compressing speech signals there are
many techniques available. The Discrete Wavelet Transform
(DWT) is a special case of the Wavelet Transform that
provides a compact representation of a signal. An important
advantage of using wavelets for speech coding is that the
compression ratio can easily be varied, while most other
techniques have fixed compression ratios. It is mainly
introduced to represent non-stationary signals more
effectively than Fourier transform and since it retains both
the time and frequency aspect of the signal. The main
advantage of the wavelet transforms is that it has a varying
window size, being broad at low frequencies and narrow at
high frequencies, thus leading to an optimal time frequency
resolution in all frequency ranges [5].
The DWT is computed by successive low pass filtering
and high pass filtering of the discrete time domain signal
called the Mallat algorithm [6]. The DWT of the original
signal can be obtained by concatenating the low frequency
components a[n] and high frequency components d[n],
starting from the last level of decomposition. It uses multi
resolution filter banks for the analysis and it is defined by the
following equation
W ( J , K ) = ¦ J ¦k X ( k ) 2 − J / 2ψ ( 2 − J N − K ) (1)
Where ȥ (t) is the basic analyzing function used and it is
called as mother wavelet [7]. Here the time-scale
representation of a signal can be obtained using digital
filtering techniques. The filtering and decimation process is
continued until the desired level is reached. The successive
2012 World Congress on Information and Communication Technologies
high pass and low pass filtering of the signal can be
represented using the following equations
Yhigh [k ] = ¦ n x[n]g[2k − n] (2)
Ylow [k ] = ¦ n x[n]h[2k − n] (3)
Where Y high and Y low are the outputs of the high pass
and low pass filters obtained by sub sampling by 2
[8].Assembling back the processed signal into the original
signal without loss of information is called synthesis.
Whereas the mathematical manipulation that affects
synthesis is called inverse discrete wavelet transform
(IDWT). Different types of wavelets like Haar, db3, db7 and
db10 are experienced while implementation.
IV. LINEAR PREDICTIVE CODING
Linear Predictive Coding (LPC) is used to compress the
speech signal without losing its audibility. It breaks the
speech into segments and then sends the voiced or unvoiced
information, the pitch period and the coefficients for the
filter. It is one of the most powerful methods used in audio
and speech signal processing which extracts speech
parameters like pitch formants and spectra. LPC is used to
estimate basic speech parameters like pitch, formants and
spectra. The principle behind the use of LPC is to minimize
the sum of the squared differences between the original
speech and estimated speech signal over a finite duration [9].
It allows encoding of good quality speech at a low bit rate
and it also provides extremely accurate estimates of speech
parameters. The most important aspect of LPC is the linear
predictive filter which allows the value of the next sample to
be determined by a linear combination of previous samples
[7].The predictor coefficients are represented by ak and it is
normally estimated in every frame with size of 20 ms long.
Another important parameter is the gain (G). The transfer
function of the time varying digital filter is given by [10]
G
H ( z) = (4)
1 − ¦ a k z −k
The summation is computed starting at k=1 up to p,
which will be 10 for the LPC-10 algorithm. This means that
only the first 10 coefficient are transmitted to the LPC
synthesizer. Hence guaranteeing the stability of system H (z),
Levinson – Durbin recursion will be utilized to compute the
required parameters for the auto-correlation method [13].
The LPC analysis of each frame also involves the decision
making process of concluding if a sound is voiced or
unvoiced. A pitch detecting algorithm is employed to
determine the correct pitch period or the frequency. The
967
performance of the compression techniques LPC and DWT
used for the datasets are illustrated in “fig 5”.Here nine
samples are used for implementation, in which the five
samples are given in this paper.
V. PERFORMANCE EVALUATION
The performance of the compression is analyzed by
comparing the compressed and decompressed signal against
the original signal [11].There are two ways in which the
techniques can be evaluated such as objectively and
subjectively. Objective analysis is done by evaluating the
performance of parameters such as Compression Ratio (CR),
Peak Signal to Noise ratio (PSNR), and Normalized Root
Mean Square Error Rate (NRMSE) [11].Whereas subjective
Based on the performance evaluation, the DWT gives
higher PSNR and lower NRMSE than the LPC. In the
Compression ratio also the DWT produces good quality than
the other method. The table1 shows the Performance
Analysis for the DWT and LPC based on PSNR, NRMSE
and CR. The following “fig 2, 3 and 4” represents the
performance of two techniques in the graphical form.
Performance analysis
based on CR
0.008
0.006

Range
analysis is based on hearing the reconstructed signal and 0.004
LPC
making the judgment which is done by Mean Opinion Score DWT
0.002
(MOS) [12]. For calculating the performance nine speech
0
samples are taken from various speakers and each file has a
1 2 3 4 5 6 7 8 9
different size with respect to other files. In this paper the
objective analysis is done in order to evaluate the parameters Number of samples
and the formulas are given below
Figure 2. Performance evaluation based on CR
A. Compression Ratio(CR)

Length( x(n)) Performance analysis

CR = (5) based on PSNR
Length(r (n))
16
The compression ratio can be calculated using the above
formula, where x(n) is the original signal and r(n) is the 15
Range

LPC
reconstructed signal [2] 14
DWT
13
B. Peak Signal to Noise Ratio (PSNR) 12
1 2 3 4 5 6 7 8 9
2 Number of samples
( NX )
PSNR = 10 log10 2
(6)
X −r Figure 3. Performance evaluation based on PSNR

The PSNR can be calculated using the above formula, Performance analysis based on
where N is length of the reconstructed signal, X is the
maximum absolute square value of the signal x and || x-r | |2 NRMSE
is the energy of the difference between the original and
reconstructed signals [12] 0.008
0.006
Range

LPC
0.004
C. Normalized Root Mean Square Error (NRMSE) 0.002
DWT

0
2 1 2 3 4 5 6 7 8 9
( x(n) − r (n))
NRMSE = (7) Number of samples
( x(n) − μx(n)) 2
Figure 4. Performance evaluation based on NRMSE
The NRMSE can be calculated using the above formula,
where x (n) is the speech signal, r(n) is the reconstructed
signal and x(n) is the mean of the speech signal [7].

968 2012 World Congress on Information and Communication Technologies

 TABLE I. PERFORMANCE ANALYSIS USING DWT AND LPC
Speech sample DWT LPC

PSNR NRMSE CR PSNR NRMSE CR

Friday Rocks 15.18606 0.00255 0.9684 14.57136 0.00364 1.0146
Have a Good Weekend 14.84219 0.00311 0.3684 14.21918 0.00446 1.0244
Get Away From Me 14.26461 0.00434 0.982 13.66169 0.00614 1.0010
Water 15.19349 0.00254 0.7807 14.58647 0.00361 1.0057
Do U Like It 14.64838 0.00348 0.6150 14.04234 0.00493 1.0046
Dinner is Served 15.35273 0.00232 0.7748 14.74981 0.00328 1.0010
Come On You Can Do It 14.54993 0.00368 0.6280 13.92235 0.00529 1.0298
Can You Keep a Secret 14.93618 0.00295 0.8096 14.32655 0.00419 1.0087
Am I Totally Screwed Or 15.57102 0.00204 0.6147 14.96809 0.00289 1.0010

[3] D.Ambika,V.Radha,”Secure Speech Communication – A Review”,

VI. CONCLUSION
Data compression is the technology of representing
information with lowest number of bits. In general a good
reconstructed signal should produce high PSNR and low
NRMSE which means the signal have low error and high
reliability. In accordance with compression based algorithms
using LPC and DWT, the DWT performs very well in
processing which achieves higher PSNR and lower NRMSE.
The compression ratio provided by DWT also is better than
the LPC. But if the wavelet like db3, haar, db5 and db10 are
used the compression ratio differs accordingly. If frame size
is larger LPC technique is not able to analyze localized
events accurately. The Wavelet transforms proved to be a
useful tool for analysis of non stationary signals. It uses short
windows for high frequencies and long window at low
frequencies. This results in multi-resolutions analysis by
which the signal is analyzed with different resolutions at
different frequencies.
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2012 World Congress on Information and Communication Technologies 969

Samples Original signal Reconstructed signal using LPC Reconstructed signal using DWT

Figure 5. Performance of the compression techniques using LPC and DWT

970 2012 World Congress on Information and Communication Technologies