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Adaptive Algorithm For Speech Compression Using Cosine Packet Transform
Adaptive Algorithm For Speech Compression Using Cosine Packet Transform
Adaptive Algorithm For Speech Compression Using Cosine Packet Transform
I. INTRODUCTION
With rapid deployment of speech compression technologies,
more and more speech content is stored and transmitted in
compressed formats. Speech signals has unique properties that
differ from a general audio/music signals. First, speech is a
signal that is more structured and band-limited around 4 kHz.
These two facts can be exploited through different models and
approaches and at the end, make it easier to compress. Today,
applications of speech compression involve real time
processing in mobile satellite communications, cellular
telephony, internet telephony, audio for videophones or video
teleconferencing systems, among others. Other applications
include also storage and synthesis systems used, for example,
in voice mail systems, voice memo wristwatches, voice logging
recorders and interactive PC software[1]. The idea of speech
compression is to compress speech signal to take up less
storage space and less bandwidth for transmission. To meet this
goal different methods for compression have been designed
and developed by various researchers [2-7]. The speech
compression is used in digital telephony, in multimedia and in
the security of digital communications. Before the introduction
of Packet based transform techniques, audio coding techniques
used DFT and DCT with window functions such as rectangular
and sine-taper functions. However, these early coding
techniques have failed to fulfil the contradictory requirements
imposed by high-quality audio coding. For example, with a
rectangular window the analysis/synthesis system is critically
sampled, i.e., the overall number of the transformed domain
samples is equal to the number of time domain samples, but the
system suffers from poor frequency resolution and block
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x(t )
Ai cos T i (t )
(1)
i 1
A cos Z t
x s (t )
(2)
i 1
Ex
| A |
i
(3)
i l
Ci D i
N 1
S (2 x 1)i
,
2 N
f ( x) cos
x 0
(4)
for i = 0,1,2,,N1.
Where
Di
2
N
for i 0
(5)
for i z 0
Computation of
DCT
C (i
0)
1
N
N 1
f ( x)
Extracting the
coefficients (Ci)
x 0
Adaptive Threshold
Detector
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a . Ex
N
Coefficients from
DCT (Ci)
Initialize
b=-10log (a), a<1
Compute the
Threshold (t)
(7)
b 10. log10 a
(8)
an
b
10
10
tn
Ex
N
YES
STOP
If
t > Ci
(9)
YES
If
Ex > b
(10)
Ci= Ci
Ci= 0
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DWT
Compression Ratio
50
DCT
40
WPT
30
Proposed
Algorithm
20
10
0
1
DCT
WPT
6.1229
6.1462
6.1462
6.1473
6.1452
6.1482
6.1473
6.1482
6.1482
6.1482
6.1479
6.1337
6.1477
6.1272
6.1461
6.1468
6.1461
6.1482
6.1452
6.1443
6.1421
6.1421
6.1421
6.1421
6.1421
6.1421
6.1421
6.1421
6.1421
6.1421
6.1421
6.1421
6.1421
6.1421
6.1421
6.1421
6.1421
6.1421
6.1421
6.1421
11.8444
12.1766
12.4397
12.3433
42.1520
51.5191
23.8063
40.9006
26.5968
35.1952
19.9917
21.5817
13.8164
30.9609
15.5718
30.4582
22.4461
29.2490
26.8928
60.5990
Proposed
Adaptive
algorithm
11.7985
12.2632
11.1433
12.8633
45.8856
55.7188
23.9820
43.0466
28.5052
36.0922
20.7104
21.7237
13.9029
31.1392
15.8481
31.6369
23.2086
30.8083
27.1709
68.0507
10
DWT
4
5
6
7
Speech Signal Sam ple
DCT
50
WPT
40
Proposed
Algorithm
30
20
10
0
1
10
DCT
WPT
Proposed
Adaptive
algorithm
6.144
6.142
27.027
28.275
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REFERENCES
[1]. R. W. Yeung, A First Course in Information Theory, New York:
Kluwer Academic/Plenum Publishers, 2002.
[2]. A.Gersho, Advances in Speech and Video Compressions,
Proceedings of the IEEE, vol. 82, pp. 900-918, June 1994.
[3]. J.L.Flanagaran,
M.R.Schroeder,
B.S.Atal,
R.E.Crocherie,
N.S.Jayant and J.M.Tribolet, Speech Coding, IEEE Transactions
on Communications, vol. 27, pp.710-737, April 1979.
[4]. P.Noll, Wideband Speech and Audio Coding, IEEE
Communications Magazine, pp. 34-44, Nov. 1993.
[5]. K. Sayood and J. C. Borkenhagen, Use of residual redundancy in
the design of joint source/channel coders, IEEE Transactions on
Communications, 39(6):838-846, June 1991.
[6]. Edler, B., Coding of Audio Signals with Overlapping Block
Transform and Adaptive Window Functions, (in German),
Frequenz, vol.43, pp.252-256, 1989.
[7]. Q. Memon, T. Kasparis, Transform Coding of Signals Using
Approximate Trigonometric Expansions. Journal of Electronic
Imaging, Vol. 6, No. 4, October 1997, pp. 494-503.
[8]. C. E. Shannon, .A mathematical theory of communications,. Bell
System Technical Journal, vol. 27, pp. 379.423, 623.656, 1948.
[9]. A. N. Kolmogorov, .On the Shannon theory of information
transmission in the case of continuous signals,. Trans. IRE, vol. IT2, pp. 102.108, 1956.
[10]. D. L. Donoho, M. Vetterli, R. A. Devore, and I. Daubechies, .Data
compression and harmonic analysis,. IEEE Trans. Inf. Theory, vol.
44, no. 6, pp. 2435.2476, 1998.
[11]. N. Ahmed, T. Natarajan, and K. R. Rao, Discrete cosine
transform, IEEE Transactions on Computers, vol. C-32, pp. 90-93,
Jan. 1974.
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