Abstract: This paper proposes strategies in level crossing (LC) sampling and reconstruction that provide alias-free high-fidelity signal reconstruction for speech signals without exponentially increasing sample number with increasing bit-depth. We introduce methods in LC sampling that reduce the sampling rate close to the Nyquist frequency even for large bit-depth. The results indicate that larger variation in the sampling intervals leads to alias-free sampling scheme; this is achieved by either reducing the bit-depth or adding a jitter to the system for high bit-depths. In conjunction with windowing, the signal is reconstructed from the LC samples using an efficient Toeplitz reconstruction algorithm.
Abstract: The telemedicine services require correct computing resource management to guarantee productivity and efficiency for medical and non-medical staff. The aim of this study was to examine web management strategies to ensure the availability of resources and services in telemedicine so as to provide medical information management with an accessible strategy. In addition, to evaluate the quality-of-service parameters, the followings were measured: delays, throughput, jitter, latency, available bandwidth, percent of access and denial of services based of web management performance map with profiles permissions and database management. Through 24 different test scenarios, the results show 100% in availability of medical information, in relation to access of medical staff to web services, and quality of service (QoS) of 99% because of network delay and performance of computer network. The findings of this study suggest that the proposed strategy of web management is an ideal solution to guarantee the availability, reliability, and accessibility of medical information. Finally, this strategy offers seven user profile used at telemedicine center of Bogota-Colombia keeping QoS parameters suitable to telemedicine services.
Abstract: Telemedicine services use a large amount of data, most of which are diagnostic images in Digital Imaging and Communications in Medicine (DICOM) and Health Level Seven (HL7) formats. Metadata is generated from each related image to support their identification. This study presents the use of decision trees for the optimization of information search processes for diagnostic images, hosted on the cloud server. To analyze the performance in the server, the following quality of service (QoS) metrics are evaluated: delay, bandwidth, jitter, latency and throughput in five test scenarios for a total of 26 experiments during the loading and downloading of DICOM images, hosted by the telemedicine group server of the Universidad Militar Nueva Granada, Bogotá, Colombia. By applying decision trees as a data mining technique and comparing it with the sequential search, it was possible to evaluate the search times of diagnostic images in the server. The results show that by using the metadata in decision trees, the search times are substantially improved, the computational resources are optimized and the request management of the telemedicine image service is improved. Based on the experiments carried out, search efficiency increased by 45% in relation to the sequential search, given that, when downloading a diagnostic image, false positives are avoided in management and acquisition processes of said information. It is concluded that, for the diagnostic images services in telemedicine, the technique of decision trees guarantees the accessibility and robustness in the acquisition and manipulation of medical images, in improvement of the diagnoses and medical procedures in patients.
Abstract: In telemedicine, the image repository service is important to increase the accuracy of diagnostic support of medical personnel. This study makes comparison between two routing algorithms regarding the quality of service (QoS), to be able to analyze the optimal performance at the time of loading and/or downloading of medical images. This study focused on comparing the performance of Tabu Search with other heuristic and metaheuristic algorithms that improve QoS in telemedicine services in Colombia. For this, Tabu Search and Simulated Annealing heuristic algorithms are chosen for their high usability in this type of applications; the QoS is measured taking into account the following metrics: Delay, Throughput, Jitter and Latency. In addition, routing tests were carried out on ten images in digital image and communication in medicine (DICOM) format of 40 MB. These tests were carried out for ten minutes with different traffic conditions, reaching a total of 25 tests, from a server of Universidad Militar Nueva Granada (UMNG) in Bogotá-Colombia to a remote user in Universidad de Santiago de Chile (USACH) - Chile. The results show that Tabu search presents a better QoS performance compared to Simulated Annealing, managing to optimize the routing of medical images, a basic requirement to offer diagnostic images services in telemedicine.
Abstract: Quality of Experience (QoE) drives churn in the broadband networks industry, and good QoE plays a large part in the retention of customers. QoE is known to be affected by the Quality of Service (QoS) factors packet loss probability (PLP), delay and delay jitter caused by the network. Earlier results have shown that the relationship between these QoS factors and QoE is non-linear, and may vary from application to application. We use the network emulator Netem as the basis for experimentation, and evaluate how QoE varies as we change the emulated QoS metrics. Focusing on Video-on-Demand, we discovered that the reported QoE may differ widely for users of different age groups, and that the most demanding age group (the youngest) can require an order of magnitude lower PLP to achieve the same QoE than is required by the most widely studied age group of users. We then used a bottleneck TCP model to evaluate the capacity cost of achieving an order of magnitude decrease in PLP, and found it be (almost always) a 3-fold increase in link capacity that was required.
Abstract: This paper presents experimental results on testing the symbol-error-rate (SER) performance of quadrature amplitude modulation (QAM) systems employing symmetric pulse-shaping square-root (SR) filters designed by minimizing the roughness function and by minimizing the peak-to-average power ratio (PAR). The device used in the experiments is the 'bladeRF' software-defined-radio platform. PAR is a well-known measurement, whereas the roughness function is a concept for measuring the jitter-induced interference. The experimental results show that the system employing minimum-roughness pulse-shaping SR filters outperforms the system employing minimum-PAR pulse-shaping SR filters in the sense of SER performance.
Abstract: Worldwide Interoperability for Microwave Access, is a broadband technology, which can effectively transmit a data across a group of users using Multicast and Broadcast Service. WiMAX belongs to a family of (IEEE 802.16) standards and is evolving as a fourth generation technology. WiMAX is the next generation technology that offers wireless access over long distances. MBS zone, which is a group of base stations that are broadcasting the same multicast packets which defines Multicast and Broadcast services. Handover is a process of transferring an ongoing call or data session from one channel connected to the core network to another channel. The handover causes authentication, delay, packet loss, jitter that mainly affects the communication. In this paper, we present a survey on handover security issues in WiMAX.
Abstract: The use of wireless technology in industrial networks
has gained vast attraction in recent years. In this paper, we have
thoroughly analyzed the effect of contention window (CW) size on
the performance of IEEE 802.11-based industrial wireless networks
(IWN), from delay and reliability perspective. Results show that the
default values of CWmin, CWmax, and retry limit (RL) are far from
the optimum performance due to the industrial application
characteristics, including short packet and noisy environment. In this
paper, an adaptive CW algorithm (payload-dependent) has been
proposed to minimize the average delay. Finally a simple, but
effective CW and RL setting has been proposed for industrial
applications which outperforms the minimum-average-delay solution
from maximum delay and jitter perspective, at the cost of a little
higher average delay. Simulation results show an improvement of up
to 20%, 25%, and 30% in average delay, maximum delay and jitter
respectively.
Abstract: Based on application requirements, nodes are static or
mobile in Wireless Sensor Networks (WSNs). Mobility poses
challenges in protocol design, especially at the link layer requiring
mobility adaptation algorithms to localize mobile nodes and predict
link quality to be established with them. This study implements
XMAC and Berkeley Media Access Control (BMAC) routing
protocols to evaluate performance under WSN’s static and mobility
conditions. This paper gives a comparative study of mobility-aware
MAC protocols. Routing protocol performance, based on Average
End to End Delay, Average Packet Delivery Ratio, Average Number
of hops, and Jitter is evaluated.
Abstract: IEEE 802.16 (WiMAX) aims to present high speed
wireless access to cover wide range coverage. The base station (BS)
and the subscriber station (SS) are the main parts of WiMAX.
WiMAX uses either Point-to-Multipoint (PMP) or mesh topologies.
In the PMP mode, the SSs connect to the BS to gain access to the
network. However, in the mesh mode, the SSs connect to each other
to gain access to the BS.
The main components of QoS management in the 802.16 standard
are the admission control, buffer management and packet scheduling.
In this paper, we use QualNet 5.0.2 to study the performance of
different scheduling schemes, such as WFQ, SCFQ, RR and SP when
the numbers of SSs increase. We find that when the number of SSs
increases, the average jitter and average end-to-end delay is increased
and the throughput is reduced.
Abstract: In this paper, we study statistical multiplexing of VBR
video in ATM networks. ATM promises to provide high speed realtime
multi-point to central video transmission for telemedicine
applications in rural hospitals and in emergency medical services.
Video coders are known to produce variable bit rate (VBR) signals
and the effects of aggregating these VBR signals need to be
determined in order to design a telemedicine network infrastructure
capable of carrying these signals. We first model the VBR video
signal and simulate it using a generic continuous-data autoregressive
(AR) scheme. We carry out the queueing analysis by the Fluid
Approximation Model (FAM) and the Markov Modulated Poisson
Process (MMPP). The study has shown a trade off: multiplexing
VBR signals reduces burstiness and improves resource utilization,
however, the buffer size needs to be increased with an associated
economic cost. We also show that the MMPP model and the Fluid
Approximation model fit best, respectively, the cell region and the
burst region. Therefore, a hybrid MMPP and FAM completely
characterizes the overall performance of the ATM statistical
multiplexer. The ramifications of this technology are clear: speed,
reliability (lower loss rate and jitter), and increased capacity in video
transmission for telemedicine. With migration to full IP-based
networks still a long way to achieving both high speed and high
quality of service, the proposed ATM architecture will remain of
significant use for telemedicine.
Abstract: The goal of speech parameterization is to extract the relevant information about what is being spoken from the audio signal. In speech recognition systems Mel-Frequency Cepstral Coefficients (MFCC) and Relative Spectral Mel-Frequency Cepstral Coefficients (RASTA-MFCC) are the two main techniques used. It will be shown in this paper that it presents some modifications to the original MFCC method. In our work the effectiveness of proposed changes to MFCC called Modified Function Cepstral Coefficients (MODFCC) were tested and compared against the original MFCC and RASTA-MFCC features. The prosodic features such as jitter and shimmer are added to baseline spectral features. The above-mentioned techniques were tested with impulsive signals under various noisy conditions within AURORA databases.
Abstract: This paper presents a new strategy of identification
and classification of pathological voices using the hybrid method
based on wavelet transform and neural networks. After speech
acquisition from a patient, the speech signal is analysed in order to
extract the acoustic parameters such as the pitch, the formants, Jitter,
and shimmer. Obtained results will be compared to those normal and
standard values thanks to a programmable database. Sounds are
collected from normal people and patients, and then classified into
two different categories. Speech data base is consists of several
pathological and normal voices collected from the national hospital
“Rabta-Tunis". Speech processing algorithm is conducted in a
supervised mode for discrimination of normal and pathology voices
and then for classification between neural and vocal pathologies
(Parkinson, Alzheimer, laryngeal, dyslexia...). Several simulation
results will be presented in function of the disease and will be
compared with the clinical diagnosis in order to have an objective
evaluation of the developed tool.
Abstract: This paper study the high-level modelling and design
of delta-sigma (ΔΣ) noise shapers for audio Digital-to-Analog
Converter (DAC) so as to eliminate the in-band Signal-to-Noise-
Ratio (SNR) degradation that accompany one channel mismatch in
audio signal. The converter combines a cascaded digital signal
interpolation, a noise-shaping single loop delta-sigma modulator with
a 5-bit quantizer resolution in the final stage. To reduce sensitivity of
Digital-to-Analog Converter (DAC) nonlinearities of the last stage, a
high pass second order Data Weighted Averaging (R2DWA) is
introduced. This paper presents a MATLAB description modelling
approach of the proposed DAC architecture with low distortion and
swing suppression integrator designs. The ΔΣ Modulator design can
be configured as a 3rd-order and allows 24-bit PCM at sampling rate
of 64 kHz for Digital Video Disc (DVD) audio application. The
modeling approach provides 139.38 dB of dynamic range for a 32
kHz signal band at -1.6 dBFS input signal level.
Abstract: Phase locked loops in 10 Gb/s and faster data links are
low phase noise devices. Characterization of their phase jitter
transfer functions is difficult because the intrinsic noise of the PLLs
is comparable to the phase noise of the reference clock signal. The
problem is solved by using a linear model to account for the intrinsic
noise. This study also introduces a novel technique for measuring the
transfer function. It involves the use of the reference clock as a
source of wideband excitation, in contrast to the commonly used
sinusoidal excitations at discrete frequencies. The data reported here
include the intrinsic noise of a PLL for 10 Gb/s links and the jitter
transfer function of a PLL for 12.8 Gb/s links. The measured transfer
function suggests that the PLL responded like a second order linear
system to a low noise reference clock.
Abstract: In this paper is to evaluate audio and speech quality
with the help of Digital Audio Watermarking Technique under the
different types of attacks (signal impairments) like Gaussian Noise,
Compression Error and Jittering Effect. Further attacks are
considered as Hostile Environment. Audio and Speech Quality
Evaluation is an important research topic. The traditional way for
speech quality evaluation is using subjective tests. They are reliable,
but very expensive, time consuming, and cannot be used in certain
applications such as online monitoring. Objective models, based on
human perception, were developed to predict the results of subjective
tests. The existing objective methods require either the original
speech or complicated computation model, which makes some
applications of quality evaluation impossible.
Abstract: Emotion recognition is an important research field that finds lots of applications nowadays. This work emphasizes on recognizing different emotions from speech signal. The extracted features are related to statistics of pitch, formants, and energy contours, as well as spectral, perceptual and temporal features, jitter, and shimmer. The Artificial Neural Networks (ANN) was chosen as the classifier. Working on finding a robust and fast ANN classifier suitable for different real life application is our concern. Several experiments were carried out on different ANN to investigate the different factors that impact the classification success rate. Using a database containing 7 different emotions, it will be shown that with a proper and careful adjustment of features format, training data sorting, number of features selected and even the ANN type and architecture used, a success rate of 85% or even more can be achieved without increasing the system complicity and the computation time
Abstract: Routing in mobile ad hoc networks is a challenging task because nodes are free to move randomly. In DSR like all On- Demand routing algorithms, route discovery mechanism is associated with great delay. More Clearly in DSR routing protocol to send route reply packet, when current route breaks, destination seeks a new route. In this paper we try to change route selection mechanism proactively. We also define a link stability parameter in which a stability value is assigned to each link. Given this feature, destination node can estimate stability of routes and can select the best and more stable route. Therefore we can reduce the delay and jitter of sending data packets.
Abstract: The voice signal in Voice over Internet protocol (VoIP) system is processed through the best effort policy based IP network, which leads to the network degradations including delay, packet loss jitter. The work in this paper presents the implementation of finite impulse response (FIR) filter for voice quality improvement in the VoIP system through distributed arithmetic (DA) algorithm. The VoIP simulations are conducted with AMR-NB 6.70 kbps and G.729a speech coders at different packet loss rates and the performance of the enhanced VoIP signal is evaluated using the perceptual evaluation of speech quality (PESQ) measurement for narrowband signal. The results show reduction in the computational complexity in the system and significant improvement in the quality of the VoIP voice signal.
Abstract: In this paper, an ultra low power and low jitter 12bit
CMOS digitally controlled oscillator (DCO) design is presented.
Based on a ring oscillator implemented with low power Schmitt
trigger based inverters. Simulation of the proposed DCO using 32nm
CMOS Predictive Transistor Model (PTM) achieves controllable
frequency range of 550MHz~830MHz with a wide linearity and high
resolution. Monte Carlo simulation demonstrates that the time-period
jitter due to random power supply fluctuation is under 31ps and the
power consumption is 0.5677mW at 750MHz with 1.2V power
supply and 0.53-ps resolution. The proposed DCO has a good
robustness to voltage and temperature variations and better linearity
comparing to the conventional design.