The report is confidential, and the referee must email the report directly to Deakin University. Research degree academic referee report (DOC) KB. Research degree academic referee report (PDF) KB. This report is to provide information on the applicant’s eligibility and readiness for a research degree at Deakin University A thesis, or dissertation (abbreviated diss.), is a document submitted in support of candidature for an academic degree or professional qualification presenting the author's research and findings. In some contexts, the word "thesis" or a cognate is used for part of a bachelor's or master's course, while "dissertation" is normally applied to a doctorate.. This is the typical arrangement in Speech recognition is an interdisciplinary subfield of computer science and computational linguistics that develops methodologies and technologies that enable the recognition and translation of spoken language into text by computers. It is also known as automatic speech recognition (ASR), computer speech recognition or speech to text (STT).It incorporates knowledge and research in the computer
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Speech recognition is an interdisciplinary subfield of computer science and computational linguistics that develops methodologies and technologies that enable the recognition and translation of spoken language into text by computers. It is also known as automatic speech recognition ASRcomputer speech recognition or speech to text STT. It incorporates knowledge and research in the computer sciencelinguistics and computer engineering fields. Some speech recognition systems require "training" also called "enrollment" where an individual speaker reads text or isolated vocabulary into the system.
The system analyzes the person's specific voice and uses it to fine-tune the recognition of that person's speech, resulting in increased accuracy. Systems that do not use training are called "speaker-independent" [1] systems.
Thesis report master arabic word recognition that use training are called "speaker dependent". Speech recognition applications include voice user interfaces such as voice dialing e. find a podcast where particular words were spokensimple data entry e.
a radiology reportdetermining speaker characteristics, [2] speech-to-text processing e. The term voice recognition [3] [4] [5] or speaker identification [6] [7] [8] refers to identifying the speaker, rather than what they are saying. Recognizing the speaker can simplify the task of translating speech in systems that have been trained on a specific person's voice or it can be used to authenticate or verify the identity of a speaker as part of a security process, thesis report master arabic word recognition.
From the technology perspective, speech recognition has a long history with several waves of major innovations. Most recently, the field has benefited from advances in deep learning and big data. The advances are evidenced not only by the surge of academic papers published in the field, but more importantly by the worldwide industry adoption of a variety of deep learning methods in designing and deploying speech recognition systems. Raj Reddy was the first person to take on continuous speech recognition as a graduate student at Stanford University in the late s.
Previous systems required users to pause after each word. Reddy's system issued spoken commands for playing chess. Around this time Soviet researchers invented the dynamic time warping DTW algorithm and used it to create a recognizer capable of operating on a word vocabulary. Although DTW would be superseded by later algorithms, the technique carried on.
Achieving speaker independence remained unsolved at this time period. During the late s Leonard Baum developed the mathematics of Markov chains at the Institute for Defense Analysis. A decade later, at CMU, Raj Reddy's students James Baker and Janet M. Baker began using the Hidden Markov Model HMM for speech recognition.
The s also saw the introduction of the n-gram language model. Much of the progress in the field is owed to the rapidly increasing capabilities of computers. At the end of the DARPA program inthe best computer available to researchers was the PDP with 4 MB ram. By this point, the vocabulary of the typical commercial speech recognition system was larger than the average human vocabulary.
The Sphinx-II system was the first to do speaker-independent, large vocabulary, continuous speech recognition and it had the best performance in DARPA's evaluation. Handling continuous speech with a large vocabulary was a major milestone in the history of speech recognition. Huang went on to found the speech recognition group at Microsoft in Raj Reddy's thesis report master arabic word recognition Kai-Fu Lee joined Apple where, inhe helped develop a speech interface prototype for the Apple computer known as Casper.
Apple originally licensed software from Nuance to provide speech recognition capability to its digital assistant Siri. In the s DARPA sponsored two speech recognition programs: Effective Affordable Reusable Speech-to-Text EARS in and Global Autonomous Language Exploitation GALE, thesis report master arabic word recognition.
Four teams participated in the EARS program: IBMa team led by BBN with LIMSI and Univ. of PittsburghCambridge Universityand a team composed of ICSISRI and University of Washington. EARS funded the collection of the Switchboard telephone speech corpus containing hours of recorded conversations from over speakers. Google 's first effort at speech recognition came in after hiring some researchers from Nuance. The recordings from GOOG produced valuable data that helped Google improve their recognition systems, thesis report master arabic word recognition.
Google Voice Search is now supported in over 30 languages. In the United States, the National Security Agency has made use of a type of speech recognition for keyword spotting since at least Recordings can be indexed and analysts can run queries over the database to find conversations of interest.
Some government research programs focused on intelligence applications of speech recognition, thesis report master arabic word recognition, e. DARPA's EARS's program and IARPA 's Babel program. In the early s, speech recognition was still dominated by traditional approaches such as Hidden Markov Models combined with feedforward artificial neural networks.
AroundLSTM trained by Connectionist Temporal Classification CTC [38] started to outperform traditional speech recognition in certain applications. The use of deep feedforward non-recurrent networks for acoustic modeling was introduced during the later part of by Geoffrey Hinton and his students at the University of Toronto and by Li Deng [41] and colleagues at Microsoft Research, initially in the collaborative work between Microsoft and the University of Toronto which was subsequently expanded to include IBM and Google hence "The shared views of four research groups" subtitle in their review paper.
Researchers have begun to use deep learning techniques for language modeling as well. In the long history of speech recognition, thesis report master arabic word recognition shallow form and deep form e.
recurrent nets of artificial neural networks had been explored for many years during s, s and a few years into the s. Most speech recognition researchers who understood such barriers hence subsequently moved away from neural nets to pursue generative modeling approaches until the recent resurgence of deep learning starting around — that had overcome all these difficulties.
Hinton et al. and Deng et al. reviewed part of this recent history about how their collaboration with each other and then with colleagues across four groups University of Toronto, Microsoft, Google, and IBM ignited a renaissance of applications of deep feedforward neural networks thesis report master arabic word recognition speech recognition.
By early s speech recognition, also called voice recognition [55] [56] [57] was clearly differentiated from sp eaker recognition, and speaker independence was considered a major breakthrough, thesis report master arabic word recognition.
Until then, systems required a "training" period. A ad for a doll had carried the tagline "Finally, the doll that understands you. InMicrosoft researchers reached a historical human parity milestone of transcribing conversational telephony speech on the widely benchmarked Switchboard task. Multiple deep learning models were used to optimize speech recognition accuracy. The speech recognition word error rate was reported to be as low as 4 professional human transcribers working together on the same benchmark, which was funded by IBM Watson speech team on the same task.
Both acoustic modeling and language modeling are important parts of modern statistically-based speech recognition algorithms, thesis report master arabic word recognition.
Hidden Markov models HMMs are widely used in many systems. Language modeling is also used in many other natural language processing applications such as document classification or statistical machine translation.
Modern general-purpose speech recognition systems are based on Hidden Markov Models. These are statistical models that output a sequence of symbols or quantities. HMMs are used in speech recognition because a speech signal can be viewed as a piecewise stationary signal or a short-time stationary signal.
In a short time scale e. Speech can be thought of as a Markov model for many stochastic purposes. Another reason why HMMs are popular is that they can be trained automatically and are simple and computationally feasible to use.
In speech recognition, the hidden Markov model would output a sequence of n -dimensional real-valued vectors with n being a small integer, such as 10outputting one of these every 10 milliseconds. The vectors would consist of cepstral coefficients, which are obtained by taking a Fourier transform of a short time window of speech and decorrelating the spectrum using a cosine transformthen taking the first most significant coefficients.
The hidden Markov model will tend to have in each state a statistical distribution that is a mixture of diagonal covariance Gaussians, thesis report master arabic word recognition, which will give a thesis report master arabic word recognition for each observed vector. Each word, or for more general speech recognition systemseach phonemewill have a different output distribution; a hidden Markov model for a sequence of words or phonemes is made by concatenating the individual trained hidden Markov models for the separate words and phonemes.
Described above are the core elements of the most common, HMM-based approach to speech recognition. Modern speech recognition systems use various combinations of a number of standard techniques in order to improve results over the basic approach described above. A typical large-vocabulary system would need context dependency for the phonemes so phonemes with different left and right context have different realizations as HMM states ; it would use cepstral normalization to normalize for a different speaker and recording conditions; for further speaker normalization, it might use vocal tract length normalization VTLN for male-female normalization and maximum likelihood linear regression MLLR for more general speaker adaptation.
The features would have so-called delta and delta-delta coefficients to capture speech dynamics and in addition, might use heteroscedastic linear discriminant analysis HLDA ; or might skip the delta and delta-delta coefficients and use splicing and an LDA -based projection followed perhaps by heteroscedastic linear discriminant analysis or a global semi-tied co variance transform also known as maximum likelihood linear transformor MLLT.
Many systems use so-called discriminative training techniques that dispense with a purely statistical approach thesis report master arabic word recognition HMM parameter estimation and instead optimize some classification-related measure of the training data.
Examples are maximum mutual information MMIminimum classification error MCEand minimum phone error MPE. Decoding of the speech the term for what happens when the system is presented with a new utterance and must compute the most likely source sentence would probably use the Viterbi algorithm to find the best path, and here there is a choice between dynamically creating a combination hidden Markov model, which includes both the acoustic and language model information and combining it statically beforehand the finite state transduceror FST, approach.
A possible improvement to decoding is to keep a set of good candidates instead of just keeping the best candidate, and to use a better scoring function re scoring to rate these good candidates so that we may pick the best one according to this refined score. The set of candidates can be kept either as a list the N-best list approach or as a subset of the models a lattice.
Re scoring is usually done by trying to minimize the Bayes risk [59] or an approximation thereof : Instead of taking the source sentence with maximal probability, we try to take the sentence that minimizes the expectancy of a given loss function with regards to all possible transcriptions i.
The loss function is usually the Levenshtein distancethough it can thesis report master arabic word recognition different distances for specific tasks; the set of possible transcriptions is, of course, pruned to maintain tractability. Efficient algorithms have been devised to re score lattices represented as weighted finite state transducers with edit distances represented themselves as a finite state transducer verifying certain assumptions.
Dynamic time warping is an approach that was historically used for speech recognition but has now largely been displaced by the more successful HMM-based approach.
Dynamic time warping is an algorithm for measuring similarity between two sequences that may vary in time or speed. For instance, similarities in walking patterns would be detected, even if in one video the person was walking slowly and if in another he or she were walking more quickly, or even if there were accelerations and deceleration during the course of one observation. DTW has been applied to video, audio, and graphics — indeed, any data that can be turned into a linear representation can be analyzed with DTW.
A well-known application has been automatic speech recognition, to cope thesis report master arabic word recognition different speaking speeds. In general, it is a method that allows a computer to find an optimal match between two given sequences e. That is, the sequences are "warped" non-linearly to match each other. This sequence alignment method is often used in the context of hidden Markov models. Neural networks emerged as an attractive acoustic modeling approach in ASR in the late s.
Since then, neural networks have been used in many aspects of speech recognition such as phoneme classification, [61] phoneme classification through multi-objective evolutionary algorithms, [62] isolated word recognition, [63] audiovisual speech recognitionaudiovisual speaker recognition and speaker adaptation.
Neural networks make fewer explicit assumptions about feature statistical properties than HMMs and have several qualities making them attractive recognition models for speech recognition. When used to estimate the probabilities of a speech feature segment, neural networks allow discriminative training in a natural and efficient manner. However, in spite of their effectiveness in classifying short-time units such as individual phonemes and isolated words, [64] early neural networks were rarely successful for continuous recognition tasks because of their limited ability to model temporal dependencies.
One approach to this limitation was to use neural networks as a pre-processing, feature transformation or dimensionality reduction, [65] step prior to HMM based recognition.
Stop Beginning Your Speeches with Good Morning and Thank You and Start with This Instead
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The report is confidential, and the referee must email the report directly to Deakin University. Research degree academic referee report (DOC) KB. Research degree academic referee report (PDF) KB. This report is to provide information on the applicant’s eligibility and readiness for a research degree at Deakin University Speech recognition is an interdisciplinary subfield of computer science and computational linguistics that develops methodologies and technologies that enable the recognition and translation of spoken language into text by computers. It is also known as automatic speech recognition (ASR), computer speech recognition or speech to text (STT).It incorporates knowledge and research in the computer Sep 15, · The ATC's mission is to further Bentley’s leadership in and strategic focus on the integration of business and technology. We enrich scholarly initiatives and student learning by empowering faculty with state-of-the-art academic, information, and communication resources
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