1,402 research outputs found
An acoustic-phonetic approach in automatic Arabic speech recognition
In a large vocabulary speech recognition system the broad phonetic classification
technique is used instead of detailed phonetic analysis to overcome the variability in the
acoustic realisation of utterances. The broad phonetic description of a word is used as a
means of lexical access, where the lexicon is structured into sets of words sharing the
same broad phonetic labelling.
This approach has been applied to a large vocabulary isolated word Arabic speech
recognition system. Statistical studies have been carried out on 10,000 Arabic words
(converted to phonemic form) involving different combinations of broad phonetic
classes. Some particular features of the Arabic language have been exploited. The results
show that vowels represent about 43% of the total number of phonemes. They also show
that about 38% of the words can uniquely be represented at this level by using eight
broad phonetic classes. When introducing detailed vowel identification the percentage of
uniquely specified words rises to 83%. These results suggest that a fully detailed
phonetic analysis of the speech signal is perhaps unnecessary.
In the adopted word recognition model, the consonants are classified into four broad
phonetic classes, while the vowels are described by their phonemic form. A set of 100
words uttered by several speakers has been used to test the performance of the
implemented approach.
In the implemented recognition model, three procedures have been developed, namely
voiced-unvoiced-silence segmentation, vowel detection and identification, and automatic
spectral transition detection between phonemes within a word. The accuracy of both the
V-UV-S and vowel recognition procedures is almost perfect. A broad phonetic
segmentation procedure has been implemented, which exploits information from the
above mentioned three procedures. Simple phonological constraints have been used to
improve the accuracy of the segmentation process. The resultant sequence of labels are
used for lexical access to retrieve the word or a small set of words sharing the same broad
phonetic labelling. For the case of having more than one word-candidates, a verification
procedure is used to choose the most likely one
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
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
Computer Models for Musical Instrument Identification
PhDA particular aspect in the perception of sound is concerned with what is commonly
termed as texture or timbre. From a perceptual perspective, timbre is what allows us
to distinguish sounds that have similar pitch and loudness. Indeed most people are
able to discern a piano tone from a violin tone or able to distinguish different voices
or singers.
This thesis deals with timbre modelling. Specifically, the formant theory of timbre
is the main theme throughout. This theory states that acoustic musical instrument
sounds can be characterised by their formant structures. Following this principle, the
central point of our approach is to propose a computer implementation for building
musical instrument identification and classification systems.
Although the main thrust of this thesis is to propose a coherent and unified
approach to the musical instrument identification problem, it is oriented towards the
development of algorithms that can be used in Music Information Retrieval (MIR)
frameworks. Drawing on research in speech processing, a complete supervised system
taking into account both physical and perceptual aspects of timbre is described.
The approach is composed of three distinct processing layers. Parametric models
that allow us to represent signals through mid-level physical and perceptual representations
are considered. Next, the use of the Line Spectrum Frequencies as spectral
envelope and formant descriptors is emphasised. Finally, the use of generative and
discriminative techniques for building instrument and database models is investigated.
Our system is evaluated under realistic recording conditions using databases of isolated
notes and melodic phrases
Speaker recognition utilizing distributed DCT-II based Mel frequency cepstral coefficients and fuzzy vector quantization
In this paper, a new and novel Automatic Speaker Recognition (ASR) system is presented. The new ASR system includes novel feature extraction and vector classification steps utilizing distributed Discrete Cosine Transform (DCT-II) based Mel Frequency Cepstral Coef?cients (MFCC) and Fuzzy Vector Quantization (FVQ). The ASR algorithm utilizes an approach based on MFCC to identify dynamic features that are used for Speaker Recognition (SR)
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