Abstract: One of the main challenges in MIMO-OFDM system
to achieve the expected performances in terms of data rate
and robustness against multi-path fading channels is the channel
estimation. Several methods were proposed in the literature based on
either least square (LS) or minimum mean squared error (MMSE)
estimators. These methods present high implementation complexity
as they require the inversion of large matrices. In order to overcome
this problem and to reduce the complexity, this paper presents
a solution that benefits from the use of the STBC encoder and
transforms the channel estimation process into a set of simple
linear operations. The proposed method is evaluated via simulation
in AWGN-Rayleigh fading channel. Simulation results show a
maximum reduction of 6.85% of the bit error rate (BER) compared to
the one obtained with the ideal case where the receiver has a perfect
knowledge of the channel.
Abstract: In this paper, a Joint Source Channel coding scheme
based on LDPC codes is investigated. We consider two concatenated
LDPC codes, one allows to compress a correlated source and the
second to protect it against channel degradations. The original
information can be reconstructed at the receiver by a joint decoder,
where the source decoder and the channel decoder run in parallel by
transferring extrinsic information. We investigate the performance of
the JSC LDPC code in terms of Bit-Error Rate (BER) in the case
of transmission over an Additive White Gaussian Noise (AWGN)
channel, and for different source and channel rate parameters.
We emphasize how JSC LDPC presents a performance tradeoff
depending on the channel state and on the source correlation. We
show that, the JSC LDPC is an efficient solution for a relatively
low Signal-to-Noise Ratio (SNR) channel, especially with highly
correlated sources. Finally, a source-channel rate optimization has
to be applied to guarantee the best JSC LDPC system performance
for a given channel.
Abstract: Speaker recognition is performed in high Additive White Gaussian Noise (AWGN) environments using principals of Computational Auditory Scene Analysis (CASA). CASA methods often classify sounds from images in the time-frequency (T-F) plane using spectrograms or cochleargrams as the image. In this paper atomic decomposition implemented by matching pursuit performs a transform from time series speech signals to the T-F plane. The atomic decomposition creates a sparsely populated T-F vector in “weight space” where each populated T-F position contains an amplitude weight. The weight space vector along with the atomic dictionary represents a denoised, compressed version of the original signal. The arraignment or of the atomic indices in the T-F vector are used for classification. Unsupervised feature learning implemented by a sparse autoencoder learns a single dictionary of basis features from a collection of envelope samples from all speakers. The approach is demonstrated using pairs of speakers from the TIMIT data set. Pairs of speakers are selected randomly from a single district. Each speak has 10 sentences. Two are used for training and 8 for testing. Atomic index probabilities are created for each training sentence and also for each test sentence. Classification is performed by finding the lowest Euclidean distance between then probabilities from the training sentences and the test sentences. Training is done at a 30dB Signal-to-Noise Ratio (SNR). Testing is performed at SNR’s of 0 dB, 5 dB, 10 dB and 30dB. The algorithm has a baseline classification accuracy of ~93% averaged over 10 pairs of speakers from the TIMIT data set. The baseline accuracy is attributable to short sequences of training and test data as well as the overall simplicity of the classification algorithm. The accuracy is not affected by AWGN and produces ~93% accuracy at 0dB SNR.
Abstract: This paper describes a method for AWGN (Additive White Gaussian Noise) variance estimation in noisy stochastic signals, referred to as Multiplicative-Noising Variance Estimation (MNVE). The aim was to develop an estimation algorithm with minimal number of assumptions on the original signal structure. The provided MATLAB simulation and results analysis of the method applied on speech signals showed more accuracy than standardized AR (autoregressive) modeling noise estimation technique. In addition, great performance was observed on very low signal-to-noise ratios, which in general represents the worst case scenario for signal denoising methods. High execution time appears to be the only disadvantage of MNVE. After close examination of all the observed features of the proposed algorithm, it was concluded it is worth of exploring and that with some further adjustments and improvements can be enviably powerful.
Abstract: One of the most important challenging factors in
medical images is nominated as noise. Image denoising refers to the
improvement of a digital medical image that has been infected by
Additive White Gaussian Noise (AWGN). The digital medical image
or video can be affected by different types of noises. They are
impulse noise, Poisson noise and AWGN. Computed tomography
(CT) images are subjects to low quality due to the noise. Quality of
CT images is dependent on absorbed dose to patients directly in such
a way that increase in absorbed radiation, consequently absorbed
dose to patients (ADP), enhances the CT images quality. In this
manner, noise reduction techniques on purpose of images quality
enhancement exposing no excess radiation to patients is one the
challenging problems for CT images processing. In this work, noise
reduction in CT images was performed using two different
directional 2 dimensional (2D) transformations; i.e., Curvelet and
Contourlet and Discrete Wavelet Transform (DWT) thresholding
methods of BayesShrink and AdaptShrink, compared to each other
and we proposed a new threshold in wavelet domain for not only
noise reduction but also edge retaining, consequently the proposed
method retains the modified coefficients significantly that result good
visual quality. Data evaluations were accomplished by using two
criterions; namely, peak signal to noise ratio (PSNR) and Structure
similarity (Ssim).
Abstract: Any signal transmitted over a channel is corrupted by noise and interference. A host of channel coding techniques has been proposed to alleviate the effect of such noise and interference. Among these Turbo codes are recommended, because of increased capacity at higher transmission rates and superior performance over convolutional codes. The multimedia elements which are associated with ample amount of data are best protected by Turbo codes. Turbo decoder employs Maximum A-posteriori Probability (MAP) and Soft Output Viterbi Decoding (SOVA) algorithms. Conventional Turbo coded systems employ Equal Error Protection (EEP) in which the protection of all the data in an information message is uniform. Some applications involve Unequal Error Protection (UEP) in which the level of protection is higher for important information bits than that of other bits. In this work, enhancement to the traditional Log MAP decoding algorithm is being done by using optimized scaling factors for both the decoders. The error correcting performance in presence of UEP in Additive White Gaussian Noise channel (AWGN) and Rayleigh fading are analyzed for the transmission of image with Discrete Cosine Transform (DCT) as source coding technique. This paper compares the performance of log MAP, Modified log MAP (MlogMAP) and Enhanced log MAP (ElogMAP) algorithms used for image transmission. The MlogMAP algorithm is found to be best for lower Eb/N0 values but for higher Eb/N0 ElogMAP performs better with optimized scaling factors. The performance comparison of AWGN with fading channel indicates the robustness of the proposed algorithm. According to the performance of three different message classes, class3 would be more protected than other two classes. From the performance analysis, it is observed that ElogMAP algorithm with UEP is best for transmission of an image compared to Log MAP and MlogMAP decoding algorithms.
Abstract: The model tests were conducted in the laboratory
without and with Plastic recycled polymer in fly ash steep slopes
overlaying soft foundation soils like fly ash and powai soil in order to
check the stability of steep slope. In this experiment, fly ash is used
as a filling material and Plastic Recycled Polymers of diameter =
3mm and length = 4mm were made from waste plastic product (lower
grade plastic product). The properties of fly ash and Plastic recycled
polymers are determined. From the experiments, load and settlement
have measured. From these data, load –settlement curves have
reported. It has been observed from test results that load carrying
capacity of mixture fly ash with Plastic Recycled Polymers slope is
more than that of fly ash slope. The deformation of Plastic Recycled
Polymers slope is slightly more than that of fly ash slope. A Finite
Element Method (F.E.M.) was also evaluated using PLAXIS 3D
version. The failure pattern, deformations and factor of safety are
reported based on analytical programme. The results from
experimental data and analytical programme are compared and
reported.
Abstract: In this paper, we regard as a coded transmission over a
frequency-selective channel. We plan to study analytically the
convergence of the turbo-detector using a maximum a posteriori
(MAP) equalizer and a MAP decoder. We demonstrate that the
densities of the maximum likelihood (ML) exchanged during the
iterations are e-symmetric and output-symmetric. Under the Gaussian
approximation, this property allows to execute a one-dimensional
scrutiny of the turbo-detector. By deriving the analytical terminology
of the ML distributions under the Gaussian approximation, we confirm
that the bit error rate (BER) performance of the turbo-detector
converges to the BER performance of the coded additive white
Gaussian noise (AWGN) channel at high signal to noise ratio (SNR),
for any frequency selective channel.
Abstract: Speech enhancement is a long standing problem with
numerous applications like teleconferencing, VoIP, hearing aids and
speech recognition. The motivation behind this research work is to
obtain a clean speech signal of higher quality by applying the optimal
noise cancellation technique. Real-time adaptive filtering algorithms
seem to be the best candidate among all categories of the speech
enhancement methods. In this paper, we propose a speech
enhancement method based on Recursive Least Squares (RLS)
adaptive filter of speech signals. Experiments were performed on
noisy data which was prepared by adding AWGN, Babble and Pink
noise to clean speech samples at -5dB, 0dB, 5dB and 10dB SNR
levels. We then compare the noise cancellation performance of
proposed RLS algorithm with existing NLMS algorithm in terms of
Mean Squared Error (MSE), Signal to Noise ratio (SNR) and SNR
Loss. Based on the performance evaluation, the proposed RLS
algorithm was found to be a better optimal noise cancellation
technique for speech signals.
Abstract: The IEEE 802.22 working group aims to drive the
Digital Video Broadcasting-Terrestrial (DVB-T) bands for data
communication to the rural area without interfering the TV broadcast.
In this paper, we arrive at a closed-form expression for average
detection probability of Fusion center (FC) with multiple antenna
over the κ − μ fading channel model. We consider a centralized
cooperative multiple antenna network for reporting. The DVB-T
samples forwarded by the secondary user (SU) were combined using
Maximum ratio combiner at FC, an energy detection is performed
to make the decision. The fading effects of the channel degrades
the detection probability of the FC, a generalized independent and
identically distributed (IID) κ − μ and an additive white Gaussian
noise (AWGN) channel is considered for reporting and sensing
respectively. The proposed system performance is verified through
simulation results.
Abstract: Different order modulations combined with different
coding schemes, allow sending more bits per symbol, thus achieving
higher throughputs and better spectral efficiencies. However, it must
also be noted that when using a modulation technique such as 64-
QAM with less overhead bits, better signal-to-noise ratios (SNRs) are
needed to overcome any Inter symbol Interference (ISI) and maintain
a certain bit error ratio (BER). The use of adaptive modulation allows
wireless technologies to yielding higher throughputs while also
covering long distances. The aim of this paper is to implement an
Adaptive Modulation and Coding (AMC) features of the WiMAX
PHY in MATLAB and to analyze the performance of the system in
different channel conditions (AWGN, Rayleigh and Rician fading
channel) with channel estimation and blind equalization. Simulation
results have demonstrated that the increment in modulation order
causes to increment in throughput and BER values. These results
derived a trade-off among modulation order, FFT length, throughput,
BER value and spectral efficiency. The BER changes gradually for
AWGN channel and arbitrarily for Rayleigh and Rician fade
channels.
Abstract: When using modern Code Division Multiple Access (CDMA) in mobile communications, the user must be able to vary the transmission rate of users to allocate bandwidth efficiently. In this work, Orthogonal Variable Spreading Factor (OVSF) codes are used with the same principles applied in a low-rate superorthogonal turbo code due to their variable-length properties. The introduced system is the Variable Rate Superorthogonal Turbo Code (VRSTC) where puncturing is not performed on the encoder’s final output but rather before selecting the output to achieve higher rates. Due to bandwidth expansion, the codes outperform an ordinary turbo code in the AWGN channel. Simulations results show decreased performance compared to those obtained with the employment of Walsh-Hadamard codes. However, with OVSF codes, the VRSTC system keeps the orthogonality of codewords whilst producing variable rate codes contrary to Walsh-Hadamard codes where puncturing is usually performed on the final output.
Abstract: Steady incompressible couple stress fluid flow through two dimensional symmetric channel with stenosis is investigated. The flow consisting of a core region to be a couple stress fluid and a peripheral layer of plasma (Newtonian fluid). Assuming the stenosis to be mild, the equations governing the flow of the proposed model are solved using the slip boundary condition and closed form expressions for the flow characteristics (the dimensionless resistance to flow and wall shear stress at the maximum height of stenosis) are derived. The effects of various parameters on these flow variables have been studied. It is observed that the resistance to flow as well as the wall shear stress increase with the height of stenosis, viscosity ratio and Darcy number. However, the trend is reversed as the slip and the couple stress parameter increase.
Abstract: In this paper, we present a non-blind technique of
adding the watermark to the Fourier spectral components of audio
signal in a way such that the modified amplitude does not exceed the
maximum amplitude spread (MAS). This MAS is due to individual
Discrete fourier transform (DFT) coefficients in that particular frame,
which is derived from the Energy Spreading function given by
Schroeder. Using this technique one can store double the information
within a given frame length i.e. overriding the watermark on the
host of equal length with least perceptual distortion. The watermark
is uniformly floating on the DFT components of original signal.
This helps in detecting any intentional manipulations done on the
watermarked audio. Also, the scheme is found robust to various signal
processing attacks like presence of multiple watermarks, Additive
white gaussian noise (AWGN) and mp3 compression.
Abstract: In this paper we present a study of the impact of connection schemes on the performance of iterative decoding of Generalized Parallel Concatenated block (GPCB) constructed from one step majority logic decodable (OSMLD) codes and we propose a new connection scheme for decoding them. All iterative decoding connection schemes use a soft-input soft-output threshold decoding algorithm as a component decoder. Numerical result for GPCB codes transmitted over Additive White Gaussian Noise (AWGN) channel are provided. It will show that the proposed scheme is better than Hagenauer-s scheme and Lucas-s scheme [1] and slightly better than the Pyndiah-s scheme.
Abstract: We proposed a new class of asymmetric turbo encoder for 3G systems that performs well in both “water fall" and “error floor" regions in [7]. In this paper, a modified (optimal) power allocation scheme for the different bits of new class of asymmetric turbo encoder has been investigated to enhance the performance. The simulation results and performance bound for proposed asymmetric turbo code with modified Unequal Power Allocation (UPA) scheme for the frame length, N=400, code rate, r=1/3 with Log-MAP decoder over Additive White Gaussian Noise (AWGN) channel are obtained and compared with the system with typical UPA and without UPA. The performance tests are extended over AWGN channel for different frame size to verify the possibility of implementation of the modified UPA scheme for the proposed asymmetric turbo code. From the performance results, it is observed that the proposed asymmetric turbo code with modified UPA performs better than the system without UPA and with typical UPA and it provides a coding gain of 0.4 to 0.52dB.
Abstract: In the paper, a fast high-resolution range profile synthetic algorithm called orthogonal matching pursuit with sensing dictionary (OMP-SD) is proposed. It formulates the traditional HRRP synthetic to be a sparse approximation problem over redundant dictionary. As it employs a priori that the synthetic range profile (SRP) of targets are sparse, SRP can be accomplished even in presence of data lost. Besides, the computation complexity decreases from O(MNDK) flops for OMP to O(M(N + D)K) flops for OMP-SD by introducing sensing dictionary (SD). Simulation experiments illustrate its advantages both in additive white Gaussian noise (AWGN) and noiseless situation, respectively.
Abstract: Overloading is a technique to accommodate more
number of users than the spreading factor N. This is a bandwidth
efficient scheme to increase the number users in a fixed bandwidth.
One of the efficient schemes to overload a CDMA system is to use
two sets of orthogonal signal waveforms (O/O). The first set is
assigned to the N users and the second set is assigned to the
additional M users. An iterative interference cancellation technique is
used to cancel interference between the two sets of users. In this
paper, the performance of an overloading scheme in which the first N
users are assigned Walsh-Hadamard orthogonal codes and extra users
are assigned the same WH codes but overlaid by a fixed (quasi) bent
sequence [11] is evaluated. This particular scheme is called Quasi-
Orthogonal Sequence (QOS) O/O scheme, which is a part of
cdma2000 standard [12] to provide overloading in the downlink
using single user detector. QOS scheme are balance O/O scheme,
where the correlation between any set-1 and set-2 users are
equalized. The allowable overload of this scheme is investigated in
the uplink on an AWGN and Rayleigh fading channels, so that the
uncoded performance with iterative multistage interference
cancellation detector remains close to the single user bound. It is
shown that this scheme provides 19% and 11% overloading with
SDIC technique for N= 16 and 64 respectively, with an SNR
degradation of less than 0.35 dB as compared to single user bound at
a BER of 0.00001. But on a Rayleigh fading channel, the channel
overloading is 45% (29 extra users) at a BER of 0.0005, with an SNR
degradation of about 1 dB as compared to single user performance
for N=64. This is a significant amount of channel overloading on a
Rayleigh fading channel.
Abstract: The number of users supported in a DS-CDMA
cellular system is typically less than spreading factor (N), and the
system is said to be underloaded. Overloading is a technique to
accommodate more number of users than the spreading factor N. In
O/O overloading scheme, the first set is assigned to the N
synchronous users and the second set is assigned to the additional
synchronous users. An iterative multistage soft decision interference
cancellation (SDIC) receiver is used to remove high level of
interference between the two sets. Performance is evaluated in terms
of the maximum number acceptable users so that the system
performance is degraded slightly compared to the single user
performance at a specified BER. In this paper, the capacity of CDMA
based O/O overloading scheme is evaluated with SDIC receiver. It is
observed that O/O scheme using orthogonal Gold codes provides
25% channel overloading (N=64) for synchronous DS-CDMA
system on an AWGN channel in the uplink at a BER of 1e-5.For a
Rayleigh faded channel, the critical capacity is 40% at a BER of 5e-5
assuming synchronous users. But in practical systems, perfect chip
timing is very difficult to maintain in the uplink.. We have shown that
the overloading performance reduces to 11% for a timing
synchronization error of 0.02Tc for a BER of 1e-5.
Abstract: This paper presents an evaluation for a wavelet-based
digital watermarking technique used in estimating the quality of
video sequences transmitted over Additive White Gaussian Noise
(AWGN) channel in terms of a classical objective metric, such as
Peak Signal-to-Noise Ratio (PSNR) without the need of the original
video. In this method, a watermark is embedded into the Discrete
Wavelet Transform (DWT) domain of the original video frames
using a quantization method. The degradation of the extracted
watermark can be used to estimate the video quality in terms of
PSNR with good accuracy. We calculated PSNR for video frames
contaminated with AWGN and compared the values with those
estimated using the Watermarking-DWT based approach. It is found
that the calculated and estimated quality measures of the video
frames are highly correlated, suggesting that this method can provide
a good quality measure for video frames transmitted over AWGN
channel without the need of the original video.