14 research outputs found
DWT-DCT-Based Data Hiding for Speech Bandwidth Extension
The limited narrowband frequency range, about 300-3400Hz, used in telephone network channels results in less intelligible and poor-quality telephony speech. To address this drawback, a novel robust speech bandwidth extension using Discrete Wavelet Transform- Discrete Cosine Transform Based Data Hiding (DWTDCTBDH) is proposed. In this technique, the missing speech information is embedded in the narrowband speech signal. The embedded missing speech information is recovered steadily at the receiver end to generate a wideband speech of considerably better quality. The robustness of the proposed method to quantization and channel noises is confirmed by the mean square error test. The enhancement in the quality of reconstructed wideband speech of the proposed method over conventional methods is reasserted by subjective listening and objective tests
Recent Advances in Signal Processing
The signal processing task is a very critical issue in the majority of new technological inventions and challenges in a variety of applications in both science and engineering fields. Classical signal processing techniques have largely worked with mathematical models that are linear, local, stationary, and Gaussian. They have always favored closed-form tractability over real-world accuracy. These constraints were imposed by the lack of powerful computing tools. During the last few decades, signal processing theories, developments, and applications have matured rapidly and now include tools from many areas of mathematics, computer science, physics, and engineering. This book is targeted primarily toward both students and researchers who want to be exposed to a wide variety of signal processing techniques and algorithms. It includes 27 chapters that can be categorized into five different areas depending on the application at hand. These five categories are ordered to address image processing, speech processing, communication systems, time-series analysis, and educational packages respectively. The book has the advantage of providing a collection of applications that are completely independent and self-contained; thus, the interested reader can choose any chapter and skip to another without losing continuity
Media gateway utilizando um GPU
Mestrado em Engenharia de Computadores e Telemátic
Proceedings of the Second International Mobile Satellite Conference (IMSC 1990)
Presented here are the proceedings of the Second International Mobile Satellite Conference (IMSC), held June 17-20, 1990 in Ottawa, Canada. Topics covered include future mobile satellite communications concepts, aeronautical applications, modulation and coding, propagation and experimental systems, mobile terminal equipment, network architecture and control, regulatory and policy considerations, vehicle antennas, and speech compression
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Operating System Based Perceptual Evaluation of Call Quality in Radio Telecommunications Networks. Development of call quality assessment at mobile terminals using the Symbian operating system, comparison with traditional approaches and proposals for a tariff regime relating call charging to perceived speech quality.
Call quality has been crucial from the inception of telecommunication networks.
Operators need to monitor call quality from the end-user¿s perspective, in order to retain
subscribers and reduce subscriber ¿churn¿. Operators worry not only about call quality and
interconnect revenue loss, but also about network connectivity issues in areas where mobile
network gateways are prevalent. Bandwidth quality as experienced by the end-user is equally
important in helping operators to reduce churn.
The parameters that network operators use to improve call quality are mainly from the
end-user¿s perspective. These parameters are usually ASR (answer seizure ratio), PDD (postdial
delay), NER (network efficiency ratio), the number of calls for which these parameters
have been analyzed and successful calls. Operators use these parameters to evaluate and
optimize the network to meet their quality requirements.
Analysis of speech quality is a major arena for research. Traditionally, users¿ perception
of speech quality has been measured offline using subjective listening tests. Such tests are,
however, slow, tedious and costly. An alternative method is therefore needed; one that can be
automatically computed on the subscriber¿s handset, be available to the operator as well as to
subscribers and, at the same time, provide results that are comparable with conventional
subjective scores. QMeter® ¿ a set of tools for signal and bandwidth measurement that have
been developed bearing in mind all the parameters that influence call and bandwidth quality
experienced by the end-user ¿ addresses these issues and, additionally, facilitates dynamic tariff
propositions which enhance the credibility of the operator.
This research focuses on call quality parameters from the end-user¿s perspective. The
call parameters used in the research are signal strength, successful call rate, normal drop call
rate, and hand-over drop rate. Signal strength is measured for every five milliseconds of an
active call and average signal strength is calculated for each successful call. The successful call
rate, normal drop rate and hand-over drop rate are used to achieve a measurement of the overall
call quality. Call quality with respect to bundles of 10 calls is proposed.
An attempt is made to visualize these parameters for better understanding of where the
quality is bad, good and excellent. This will help operators, as well as user groups, to measure
quality and coverage.
Operators boast about their bandwidth but in reality, to know the locations where speed
has to be improved, they need a tool that can effectively measure speed from the end-user¿s
perspective. BM (bandwidth meter), a tool developed as a part of this research, measures the
average speed of data sessions and stores the information for analysis at different locations.
To address issues of quality in the subscriber segment, this research proposes the
varying of tariffs based on call and bandwidth quality. Call charging based on call quality as
perceived by the end-user is proposed, both to satisfy subscribers and help operators to improve
customer satisfaction and increase average revenue per user. Tariff redemption procedures are
put forward for bundles of 10 calls and 10 data sessions. In addition to the varying of tariffs,
quality escalation processes are proposed. Deploying such tools on selected or random samples
of users will result in substantial improvement in user loyalty which, in turn, will bring
operational and economic advantages
A MODEL FOR PREDICTING THE PERFORMANCE OF IP VIDEOCONFERENCING
With the incorporation of free desktop videoconferencing (DVC) software on the
majority of the world's PCs, over the recent years, there has, inevitably, been considerable
interest in using DVC over the Internet. The growing popularity of DVC
increases the need for multimedia quality assessment. However, the task of predicting
the perceived multimedia quality over the Internet Protocol (IP) networks is
complicated by the fact that the audio and video streams are susceptible to unique
impairments due to the unpredictable nature of IP networks, different types of task
scenarios, different levels of complexity, and other related factors. To date, a standard
consensus to define the IP media Quality of Service (QoS) has yet to be implemented.
The thesis addresses this problem by investigating a new approach to
assess the quality of audio, video, and audiovisual overall as perceived in low cost
DVC systems.
The main aim of the thesis is to investigate current methods used to assess the perceived
IP media quality, and then propose a model which will predict the quality of
audiovisual experience from prevailing network parameters.
This thesis investigates the effects of various traffic conditions, such as, packet loss,
jitter, and delay and other factors that may influence end user acceptance, when low
cost DVC is used over the Internet. It also investigates the interaction effects between
the audio and video media, and the issues involving the lip sychronisation
error. The thesis provides the empirical evidence that the subjective mean opinion
score (MOS) of the perceived multimedia quality is unaffected by lip synchronisation
error in low cost DVC systems.
The data-gathering approach that is advocated in this thesis involves both field and
laboratory trials to enable the comparisons of results between classroom-based experiments
and real-world environments to be made, and to provide actual real-world
confirmation of the bench tests. The subjective test method was employed
since it has been proven to be more robust and suitable for the research studies, as
compared to objective testing techniques.
The MOS results, and the number of observations obtained, have enabled a set of
criteria to be established that can be used to determine the acceptable QoS for given
network conditions and task scenarios. Based upon these comprehensive findings,
the final contribution of the thesis is the proposal of a new adaptive architecture
method that is intended to enable the performance of IP based DVC of a particular
session to be predicted for a given network condition
Speech Recognition
Chapters in the first part of the book cover all the essential speech processing techniques for building robust, automatic speech recognition systems: the representation for speech signals and the methods for speech-features extraction, acoustic and language modeling, efficient algorithms for searching the hypothesis space, and multimodal approaches to speech recognition. The last part of the book is devoted to other speech processing applications that can use the information from automatic speech recognition for speaker identification and tracking, for prosody modeling in emotion-detection systems and in other speech processing applications that are able to operate in real-world environments, like mobile communication services and smart homes