Rozprawy doktorskie: „Audio speech recognition”

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Miyajima, C., D. Negi, Y. Ninomiya, M. Sano, K. Mori, K. Itou, K. Takeda i Y. Suenaga. "Audio-Visual Speech Database for Bimodal Speech Recognition". INTELLIGENT MEDIA INTEGRATION NAGOYA UNIVERSITY / COE, 2005. http://hdl.handle.net/2237/10460.

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Seymour, R. "Audio-visual speech and speaker recognition". Thesis, Queen's University Belfast, 2008. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.492489.

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In this thesis, a number of important issues relating to the use of both audio and video information for speech and speaker recognition are investigated. A comprehensive comparison of different visual feature types is given, including both geometric and image transformation based features. A new geometric based method for feature extraction is described, as well as the novel use of curvelet based features. Different methods for constructing the feature vectors are compared, as well as feature vector sizes and the use of dynamic features. Each feature type is tested against three types of visual noise: compression, blurring and jitter. A novel method of integrating the audio and video information streams called the maximum stream posterior (MSP) is described. This method is tested in both speaker dependent and speaker independent audio-visual speech recognition (AVSR) systems, and is shown to be robust to noise in either the audio or video streams, given no prior knowledge of the noise. This method is then extended to form the maximum weighted stream posterior (MWSP) method. Finally, both the MSP and MWSP are tested in an audio-visual speaker recognition system (AVSpR). / Experiments using the XM2VTS database will show that both of these methods can outperform ,_.','/ standard methods in terms of recognition accuracy in situations where either stream is corrupted.

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3

Pachoud, Samuel. "Audio-visual speech and emotion recognition". Thesis, Queen Mary, University of London, 2010. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.528923.

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Matthews, Iain. "Features for audio-visual speech recognition". Thesis, University of East Anglia, 1998. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.266736.

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Kaucic, Robert August. "Lip tracking for audio-visual speech recognition". Thesis, University of Oxford, 1997. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.360392.

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Lucey, Simon. "Audio-visual speech processing". Thesis, Queensland University of Technology, 2002. https://eprints.qut.edu.au/36172/7/SimonLuceyPhDThesis.pdf.

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Speech is inherently bimodal, relying on cues from the acoustic and visual speech modalities for perception. The McGurk effect demonstrates that when humans are presented with conflicting acoustic and visual stimuli, the perceived sound may not exist in either modality. This effect has formed the basis for modelling the complementary nature of acoustic and visual speech by encapsulating them into the relatively new research field of audio-visual speech processing (AVSP). Traditional acoustic based speech processing systems have attained a high level of performance in recent years, but the performance of these systems is heavily dependent on a match between training and testing conditions. In the presence of mismatched conditions (eg. acoustic noise) the performance of acoustic speech processing applications can degrade markedly. AVSP aims to increase the robustness and performance of conventional speech processing applications through the integration of the acoustic and visual modalities of speech, in particular the tasks of isolated word speech and text-dependent speaker recognition. Two major problems in AVSP are addressed in this thesis, the first of which concerns the extraction of pertinent visual features for effective speech reading and visual speaker recognition. Appropriate representations of the mouth are explored for improved classification performance for speech and speaker recognition. Secondly, there is the question of how to effectively integrate the acoustic and visual speech modalities for robust and improved performance. This question is explored in-depth using hidden Markov model(HMM)classifiers. The development and investigation of integration strategies for AVSP required research into a new branch of pattern recognition known as classifier combination theory. A novel framework is presented for optimally combining classifiers so their combined performance is greater than any of those classifiers individually. The benefits of this framework are not restricted to AVSP, as they can be applied to any task where there is a need for combining independent classifiers.

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Eriksson, Mattias. "Speech recognition availability". Thesis, Linköping University, Department of Computer and Information Science, 2004. http://urn.kb.se/resolve?urn=urn:nbn:se:liu:diva-2651.

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This project investigates the importance of availability in the scope of dictation programs. Using speech recognition technology for dictating has not reached the public, and that may very well be a result of poor availability in today’s technical solutions.

I have constructed a persona character, Johanna, who personalizes the target user. I have also developed a solution that streams audio into a speech recognition server and sends back interpreted text. Johanna affirmed that the solution was successful in theory.

I then incorporated test users that tried out the solution in practice. Half of them do indeed claim that their usage has been and will continue to be increased thanks to the new level of availability.

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8

Rao, Ram Raghavendra. "Audio-visual interaction in multimedia". Diss., Georgia Institute of Technology, 1998. http://hdl.handle.net/1853/13349.

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Dean, David Brendan. "Synchronous HMMs for audio-visual speech processing". Thesis, Queensland University of Technology, 2008. https://eprints.qut.edu.au/17689/3/David_Dean_Thesis.pdf.

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Both human perceptual studies and automaticmachine-based experiments have shown that visual information from a speaker's mouth region can improve the robustness of automatic speech processing tasks, especially in the presence of acoustic noise. By taking advantage of the complementary nature of the acoustic and visual speech information, audio-visual speech processing (AVSP) applications can work reliably in more real-world situations than would be possible with traditional acoustic speech processing applications. The two most prominent applications of AVSP for viable human-computer-interfaces involve the recognition of the speech events themselves, and the recognition of speaker's identities based upon their speech. However, while these two fields of speech and speaker recognition are closely related, there has been little systematic comparison of the two tasks under similar conditions in the existing literature. Accordingly, the primary focus of this thesis is to compare the suitability of general AVSP techniques for speech or speaker recognition, with a particular focus on synchronous hidden Markov models (SHMMs). The cascading appearance-based approach to visual speech feature extraction has been shown to work well in removing irrelevant static information from the lip region to greatly improve visual speech recognition performance. This thesis demonstrates that these dynamic visual speech features also provide for an improvement in speaker recognition, showing that speakers can be visually recognised by how they speak, in addition to their appearance alone. This thesis investigates a number of novel techniques for training and decoding of SHMMs that improve the audio-visual speech modelling ability of the SHMM approach over the existing state-of-the-art joint-training technique. Novel experiments are conducted within to demonstrate that the reliability of the two streams during training is of little importance to the final performance of the SHMM. Additionally, two novel techniques of normalising the acoustic and visual state classifiers within the SHMM structure are demonstrated for AVSP. Fused hidden Markov model (FHMM) adaptation is introduced as a novel method of adapting SHMMs from existing wellperforming acoustic hidden Markovmodels (HMMs). This technique is demonstrated to provide improved audio-visualmodelling over the jointly-trained SHMMapproach at all levels of acoustic noise for the recognition of audio-visual speech events. However, the close coupling of the SHMM approach will be shown to be less useful for speaker recognition, where a late integration approach is demonstrated to be superior.

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Dean, David Brendan. "Synchronous HMMs for audio-visual speech processing". Queensland University of Technology, 2008. http://eprints.qut.edu.au/17689/.

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Both human perceptual studies and automaticmachine-based experiments have shown that visual information from a speaker's mouth region can improve the robustness of automatic speech processing tasks, especially in the presence of acoustic noise. By taking advantage of the complementary nature of the acoustic and visual speech information, audio-visual speech processing (AVSP) applications can work reliably in more real-world situations than would be possible with traditional acoustic speech processing applications. The two most prominent applications of AVSP for viable human-computer-interfaces involve the recognition of the speech events themselves, and the recognition of speaker's identities based upon their speech. However, while these two fields of speech and speaker recognition are closely related, there has been little systematic comparison of the two tasks under similar conditions in the existing literature. Accordingly, the primary focus of this thesis is to compare the suitability of general AVSP techniques for speech or speaker recognition, with a particular focus on synchronous hidden Markov models (SHMMs). The cascading appearance-based approach to visual speech feature extraction has been shown to work well in removing irrelevant static information from the lip region to greatly improve visual speech recognition performance. This thesis demonstrates that these dynamic visual speech features also provide for an improvement in speaker recognition, showing that speakers can be visually recognised by how they speak, in addition to their appearance alone. This thesis investigates a number of novel techniques for training and decoding of SHMMs that improve the audio-visual speech modelling ability of the SHMM approach over the existing state-of-the-art joint-training technique. Novel experiments are conducted within to demonstrate that the reliability of the two streams during training is of little importance to the final performance of the SHMM. Additionally, two novel techniques of normalising the acoustic and visual state classifiers within the SHMM structure are demonstrated for AVSP. Fused hidden Markov model (FHMM) adaptation is introduced as a novel method of adapting SHMMs from existing wellperforming acoustic hidden Markovmodels (HMMs). This technique is demonstrated to provide improved audio-visualmodelling over the jointly-trained SHMMapproach at all levels of acoustic noise for the recognition of audio-visual speech events. However, the close coupling of the SHMM approach will be shown to be less useful for speaker recognition, where a late integration approach is demonstrated to be superior.

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11

Reikeras, Helge. "Audio-visual automatic speech recognition using Dynamic Bayesian Networks". Thesis, Stellenbosch : University of Stellenbosch, 2011. http://hdl.handle.net/10019.1/6777.

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Rintala, Jonathan. "Speech Emotion Recognition from Raw Audio using Deep Learning". Thesis, KTH, Skolan för elektroteknik och datavetenskap (EECS), 2020. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-278858.

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Traditionally, in Speech Emotion Recognition, models require a large number of manually engineered features and intermediate representations such as spectrograms for training. However, to hand-engineer such features often requires both expert domain knowledge and resources. Recently, with the emerging paradigm of deep-learning, end-to-end models that extract features themselves and learn from the raw speech signal directly have been explored. A previous approach has been to combine multiple parallel CNNs with different filter lengths to extract multiple temporal features from the audio signal, and then feed the resulting sequence to a recurrent block. Also, other recent work present high accuracies when utilizing local feature learning blocks (LFLBs) for reducing the dimensionality of a raw audio signal, extracting the most important information. Thus, this study will combine the idea of LFLBs for feature extraction with a block of parallel CNNs with different filter lengths for capturing multitemporal features; this will finally be fed into an LSTM layer for global contextual feature learning. To the best of our knowledge, such a combined architecture has yet not been properly investigated. Further, this study will investigate different configurations of such an architecture. The proposed model is then trained and evaluated on the well-known speech databases EmoDB and RAVDESS, both in a speaker-dependent and speaker-independent manner. The results indicate that the proposed architecture can produce comparable results with state-of-the-art; despite excluding data augmentation and advanced pre-processing. It was reported 3 parallel CNN pipes yielded the highest accuracy, together with a series of modified LFLBs that utilize averagepooling and ReLU activation. This shows the power of leaving the feature learning up to the network and opens up for interesting future research on time-complexity and trade-off between introducing complexity in pre-processing or in the model architecture itself.
Traditionellt sätt, vid talbaserad känsloigenkänning, kräver modeller ett stort antal manuellt konstruerade attribut och mellanliggande representationer, såsom spektrogram, för träning. Men att konstruera sådana attribut för hand kräver ofta både domänspecifika expertkunskaper och resurser. Nyligen har djupinlärningens framväxande end-to-end modeller, som utvinner attribut och lär sig direkt från den råa ljudsignalen, undersökts. Ett tidigare tillvägagångssätt har varit att kombinera parallella CNN:er med olika filterlängder för att extrahera flera temporala attribut från ljudsignalen och sedan låta den resulterande sekvensen passera vidare in i ett så kallat Recurrent Neural Network. Andra tidigare studier har också nått en hög noggrannhet när man använder lokala inlärningsblock (LFLB) för att reducera dimensionaliteten hos den råa ljudsignalen, och på så sätt extraheras den viktigaste informationen från ljudet. Således kombinerar denna studie idén om att nyttja LFLB:er för extraktion av attribut, tillsammans med ett block av parallella CNN:er som har olika filterlängder för att fånga multitemporala attribut; detta kommer slutligen att matas in i ett LSTM-lager för global inlärning av kontextuell information. Så vitt vi vet har en sådan kombinerad arkitektur ännu inte undersökts. Vidare kommer denna studie att undersöka olika konfigurationer av en sådan arkitektur. Den föreslagna modellen tränas och utvärderas sedan på de välkända taldatabaserna EmoDB och RAVDESS, både via ett talarberoende och talaroberoende tillvägagångssätt. Resultaten indikerar att den föreslagna arkitekturen kan ge jämförbara resultat med state-of-the-art, trots att ingen ökning av data eller avancerad förbehandling har inkluderats. Det rapporteras att 3 parallella CNN-lager gav högsta noggrannhet, tillsammans med en serie av modifierade LFLB:er som nyttjar average-pooling och ReLU som aktiveringsfunktion. Detta visar fördelarna med att lämna inlärningen av attribut till nätverket och öppnar upp för intressant framtida forskning kring tidskomplexitet och avvägning mellan introduktion av komplexitet i förbehandlingen eller i själva modellarkitekturen.

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13

Ahmad, Nasir. "A motion based approach for audio-visual automatic speech recognition". Thesis, Loughborough University, 2011. https://dspace.lboro.ac.uk/2134/8564.

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The research work presented in this thesis introduces novel approaches for both visual region of interest extraction and visual feature extraction for use in audio-visual automatic speech recognition. In particular, the speaker‘s movement that occurs during speech is used to isolate the mouth region in video sequences and motionbased features obtained from this region are used to provide new visual features for audio-visual automatic speech recognition. The mouth region extraction approach proposed in this work is shown to give superior performance compared with existing colour-based lip segmentation methods. The new features are obtained from three separate representations of motion in the region of interest, namely the difference in luminance between successive images, block matching based motion vectors and optical flow. The new visual features are found to improve visual-only and audiovisual speech recognition performance when compared with the commonly-used appearance feature-based methods. In addition, a novel approach is proposed for visual feature extraction from either the discrete cosine transform or discrete wavelet transform representations of the mouth region of the speaker. In this work, the image transform is explored from a new viewpoint of data discrimination; in contrast to the more conventional data preservation viewpoint. The main findings of this work are that audio-visual automatic speech recognition systems using the new features extracted from the frequency bands selected according to their discriminatory abilities generally outperform those using features designed for data preservation. To establish the noise robustness of the new features proposed in this work, their performance has been studied in presence of a range of different types of noise and at various signal-to-noise ratios. In these experiments, the audio-visual automatic speech recognition systems based on the new approaches were found to give superior performance both to audio-visual systems using appearance based features and to audio-only speech recognition systems.

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Ibrahim, Zamri. "A novel lip geometry approach for audio-visual speech recognition". Thesis, Loughborough University, 2014. https://dspace.lboro.ac.uk/2134/16526.

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By identifying lip movements and characterizing their associations with speech sounds, the performance of speech recognition systems can be improved, particularly when operating in noisy environments. Various method have been studied by research group around the world to incorporate lip movements into speech recognition in recent years, however exactly how best to incorporate the additional visual information is still not known. This study aims to extend the knowledge of relationships between visual and speech information specifically using lip geometry information due to its robustness to head rotation and the fewer number of features required to represent movement. A new method has been developed to extract lip geometry information, to perform classification and to integrate visual and speech modalities. This thesis makes several contributions. First, this work presents a new method to extract lip geometry features using the combination of a skin colour filter, a border following algorithm and a convex hull approach. The proposed method was found to improve lip shape extraction performance compared to existing approaches. Lip geometry features including height, width, ratio, area, perimeter and various combinations of these features were evaluated to determine which performs best when representing speech in the visual domain. Second, a novel template matching technique able to adapt dynamic differences in the way words are uttered by speakers has been developed, which determines the best fit of an unseen feature signal to those stored in a database template. Third, following on evaluation of integration strategies, a novel method has been developed based on alternative decision fusion strategy, in which the outcome from the visual and speech modality is chosen by measuring the quality of audio based on kurtosis and skewness analysis and driven by white noise confusion. Finally, the performance of the new methods introduced in this work are evaluated using the CUAVE and LUNA-V data corpora under a range of different signal to noise ratio conditions using the NOISEX-92 dataset.

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Susman, Derya. "Turkish Large Vocabulary Continuous Speech Recognition By Using Limited Audio Corpus". Master's thesis, METU, 2012. http://etd.lib.metu.edu.tr/upload/12614207/index.pdf.

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Speech recognition in Turkish Language is a challenging problem in several perspectives. Most of the challenges are related to the morphological structure of the language. Since Turkish is an agglutinative language, it is possible to generate many words from a single stem by using suffixes. This characteristic of the language increases the out-of-vocabulary (OOV) words, which degrade the performance of a speech recognizer dramatically. Also, Turkish language allows words to be ordered in a free manner, which makes it difficult to generate robust language models. In this thesis, the existing models and approaches which address the problem of Turkish LVCSR (Large Vocabulary Continuous Speech Recognition) are explored. Different recognition units (words, morphs, stem and endings) are used in generating the n-gram language models. 3-gram and 4-gram language models are generated with respect to the recognition unit. Since the solution domain of speech recognition is involved with machine learning, the performance of the recognizer depends on the sufficiency of the audio data used in acoustic model training. However, it is difficult to obtain rich audio corpora for the Turkish language. In this thesis, existing approaches are used to solve the problem of Turkish LVCSR by using a limited audio corpus. We also proposed several data selection approaches in order to improve the robustness of the acoustic model.

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Ravindran, Sourabh. "Physiologically Motivated Methods For Audio Pattern Classification". Diss., Georgia Institute of Technology, 2006. http://hdl.handle.net/1853/14066.

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Human-like performance by machines in tasks of speech and audio processing has remained an elusive goal. In an attempt to bridge the gap in performance between humans and machines there has been an increased effort to study and model physiological processes. However, the widespread use of biologically inspired features proposed in the past has been hampered mainly by either the lack of robustness across a range of signal-to-noise ratios or the formidable computational costs. In physiological systems, sensor processing occurs in several stages. It is likely the case that signal features and biological processing techniques evolved together and are complementary or well matched. It is precisely for this reason that modeling the feature extraction processes should go hand in hand with modeling of the processes that use these features. This research presents a front-end feature extraction method for audio signals inspired by the human peripheral auditory system. New developments in the field of machine learning are leveraged to build classifiers to maximize the performance gains afforded by these features. The structure of the classification system is similar to what might be expected in physiological processing. Further, the feature extraction and classification algorithms can be efficiently implemented using the low-power cooperative analog-digital signal processing platform. The usefulness of the features is demonstrated for tasks of audio classification, speech versus non-speech discrimination, and speech recognition. The low-power nature of the classification system makes it ideal for use in applications such as hearing aids, hand-held devices, and surveillance through acoustic scene monitoring

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Dong, Junda. "Designing a Visual Front End in Audio-Visual Automatic Speech Recognition System". DigitalCommons@CalPoly, 2015. https://digitalcommons.calpoly.edu/theses/1382.

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Audio-visual automatic speech recognition (AVASR) is a speech recognition technique integrating audio and video signals as input. Traditional audio-only speech recognition system only uses acoustic information from an audio source. However the recognition performance degrades significantly in acoustically noisy environments. It has been shown that visual information also can be used to identify speech. To improve the speech recognition performance, audio-visual automatic speech recognition has been studied. In this paper, we focus on the design of the visual front end of an AVASR system, which mainly consists of face detection and lip localization. The front end is built upon the AVICAR database that was recorded in moving vehicles. Therefore, diverse lighting conditions and poor quality of imagery are the problems we must overcome. We first propose the use of the Viola-Jones face detection algorithm that can process images rapidly with high detection accuracy. When the algorithm is applied to the AVICAR database, we reach an accuracy of 89% face detection rate. By separately detecting and integrating the detection results from all different color channels, we further improve the detection accuracy to 95%. To reliably localize the lips, three algorithms are studied and compared: the Gabor filter algorithm, the lip enhancement algorithm, and the modified Viola-Jones algorithm for lip features. Finally, to increase detection rate, a modified Viola-Jones algorithm and lip enhancement algorithms are cascaded based on the results of three lip localization methods. Overall, the front end achieves an accuracy of 90% for lip localization.

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Moghimi, Amir Reza. "Array-based Spectro-temporal Masking For Automatic Speech Recognition". Research Showcase @ CMU, 2014. http://repository.cmu.edu/dissertations/334.

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Over the years, a variety of array processing techniques have been applied to the problem of enhancing degraded speech to improve automatic speech recognition. In this context, linear beamforming has long been the approach of choice, for reasons including good performance, robustness and analytical simplicity. While various non-linear techniques - typically based to some extent on the study of auditory scene analysis - have also been of interest, they tend to lag behind their linear counterparts in terms of simplicity, scalability and exibility. Nonlinear techniques are also more difficult to analyze and lack the systematic descriptions available in the study of linear beamformers. This work focuses on a class of nonlinear processing, known as time-frequency (T-F) masking - a.k.a. spectro-temporal masking { whose variants comprise a significant portion of the existing techniques. T-F masking is based on accepting or rejecting individual time-frequency cells based on some estimate of local signal quality. Analyses are developed that attempt to mirror the beam patterns used to describe linear processing, leading to a view of T-F masking as "nonlinear beamforming". Two distinct formulations of these "nonlinear beam patterns" are developed, based on different metrics of the algorithms behavior; these formulations are modeled in a variety of scenarios to demonstrate the flexibility of the idea. While these patterns are not quite as simple or all-encompassing as traditional beam patterns in microphone-array processing, they do accurately represent the behavior of masking algorithms in analogous and intuitive ways. In addition to analyzing this class of nonlinear masking algorithm, we also attempt to improve its performance in a variety of ways. Improvements are proposed to the baseline two-channel version of masking, by addressing both the mask estimation and the signal reconstruction stages; the latter more successfully than the former. Furthermore, while these approaches have been shown to outperform linear beamforming in two-sensor arrays, extensions to larger arrays have been few and unsuccessful. We find that combining beamforming and masking is a viable method of bringing the benefits of masking to larger arrays. As a result, a hybrid beamforming-masking approach, called "post-masking", is developed that improves upon the performance of MMSE beamforming (and can be used with any beamforming technique), with the potential for even greater improvement in the future.

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Zhang, Xianxian. "Robust speech processing based on microphone array, audio-visual, and frame selection for in-vehicle speech recognition and in-set speaker recognition". Diss., Connect to online resource, 2005. http://wwwlib.umi.com/cr/colorado/fullcit?p3190350.

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Goussard, George Willem. "Unsupervised clustering of audio data for acoustic modelling in automatic speech recognition systems". Thesis, Stellenbosch : University of Stellenbosch, 2011. http://hdl.handle.net/10019.1/6686.

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Thesis (MScEng (Electrical and Electronic Engineering))--University of Stellenbosch, 2011.
ENGLISH ABSTRACT: This thesis presents a system that is designed to replace the manual process of generating a pronunciation dictionary for use in automatic speech recognition. The proposed system has several stages. The first stage segments the audio into what will be known as the subword units, using a frequency domain method. In the second stage, dynamic time warping is used to determine the similarity between the segments of each possible pair of these acoustic segments. These similarities are used to cluster similar acoustic segments into acoustic clusters. The final stage derives a pronunciation dictionary from the orthography of the training data and corresponding sequence of acoustic clusters. This process begins with an initial mapping between words and their sequence of clusters, established by Viterbi alignment with the orthographic transcription. The dictionary is refined iteratively by pruning redundant mappings, hidden Markov model estimation and Viterbi re-alignment in each iteration. This approach is evaluated experimentally by applying it to two subsets of the TIMIT corpus. It is found that, when test words are repeated often in the training material, the approach leads to a system whose accuracy is almost as good as one trained using the phonetic transcriptions. When test words are not repeated often in the training set, the proposed approach leads to better results than those achieved using the phonetic transcriptions, although the recognition is poor overall in this case.
AFRIKAANSE OPSOMMING: Die doelwit van die tesis is om ’n stelsel te beskryf wat ontwerp is om die handgedrewe proses in die samestelling van ’n woordeboek, vir die gebruik in outomatiese spraakherkenningsstelsels, te vervang. Die voorgestelde stelsel bestaan uit ’n aantal stappe. Die eerste stap is die segmentering van die oudio in sogenaamde sub-woord eenhede deur gebruik te maak van ’n frekwensie gebied tegniek. Met die tweede stap word die dinamiese tydverplasingsalgoritme ingespan om die ooreenkoms tussen die segmente van elkeen van die moontlike pare van die akoestiese segmente bepaal. Die ooreenkomste word dan gebruik om die akoestiese segmente te groepeer in akoestiese groepe. Die laaste stap stel die woordeboek saam deur gebruik te maak van die ortografiese transkripsie van afrigtingsdata en die ooreenstemmende reeks akoestiese groepe. Die finale stap begin met ’n aanvanklike afbeelding vanaf woorde tot hul reeks groep identifiseerders, bewerkstellig deur Viterbi belyning en die ortografiese transkripsie. Die woordeboek word iteratief verfyn deur oortollige afbeeldings te snoei, verskuilde Markov modelle af te rig en deur Viterbi belyning te gebruik in elke iterasie. Die benadering is getoets deur dit eksperimenteel te evalueer op twee subversamelings data vanuit die TIMIT korpus. Daar is bevind dat, wanneer woorde herhaal word in die afrigtingsdata, die stelsel se benadering die akkuraatheid ewenaar van ’n stelsel wat met die fonetiese transkripsie afgerig is. As die woorde nie herhaal word in die afrigtingsdata nie, is die akkuraatheid van die stelsel se benadering beter as wanneer die stelsel afgerig word met die fonetiese transkripsie, alhoewel die akkuraatheid in die algemeen swak is.

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MORRONE, GIOVANNI. "Metodologie di Apprendimento Profondo per l'Elaborazione Audio-Video del Parlato in Ambienti Rumorosi". Doctoral thesis, Università degli studi di Modena e Reggio Emilia, 2021. http://hdl.handle.net/11380/1245516.

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Spesso la comunicazione tra persone è un'esperienza audio-visiva. Infatti, una persona ascolta le parole pronunciate da un interlocutore e contemporaneamente può anche vedere i movimenti faciali ed altri segni che possono trasmettere informazioni sul parlato. Tuttavia, la comunicazione attraverso la lingua parlata può essere influenzata negativamente da rumori di sottofondo ed artefatti, i quali sono molto comuni in ambienti reali. Recuperare il parlato ripulito a partire da sorgenti sonore degradate è fondamentale per molte applicazioni, ad esempio per il riconoscimento vocale automatico oppure per gli apparecchi acustici. La ricerca nell'ambito delle neuroscienze ha dimostrato che guardare il volto di una persona mentre sta parlando migliora la capacità umana di focalizzare l'attenzione su uno stimolo sonoro specifico, silenziando sorgenti rumorose esterne. Questa tesi ha l'obiettivo di provare a sfruttare la natura bi-modale, ovvero audio-visiva, del parlato per eseguire lo speech enhancement, il riconoscimento vocale automatico e lo speech inpainting. Iniziamo presentando un nuovo approccio per risolvere il problema di estrazione della voce di un interlocutore di interesse in uno scenario cocktail party. A differenza della grande maggioranza dei lavori precedenti, noi sfruttiamo un rilevatore pre-allenato di punti salienti facciali ed usiamo il movimento di tali punti come input video in un modello di apprendimento profondo. In questo modo, solleviamo i nostri modelli dal compito di imparare le caratteristiche visive direttamente dai pixel contenuti nei fotogrammi dei video. I nostri modelli sono allenati e testati su due dataset largamente utilizzati e di dimensione limitata, e sono in grado di eseguire lo speech enhancement in presenza di più interlocutori che parlano contemporaneamente, ed anche per persone che non sono osservate durante l'addestramento. Motivati da questi risultati, analizziamo in che modo lo speech enhancement audio-visivo può aiutare il riconoscimento vocale automatico, sfruttando un'architettura di apprendimento multi-task. Quindi, abbiamo ideato una strategia in cui la fase di addestramento dello speech enhancement è alternata con la fase di riconoscimento vocale. Osserviamo che, in generale, l’ottimizzazione congiunta delle due fasi fornisce un notevole miglioramento dell’accuratezza del riconoscimento vocale rispetto ai modelli baseline audio-visivi addestrati solamente per eseguire il riconoscimento vocale. Infine, indaghiamo se l'informazione visiva può essere utile per lo speech inpainting, ovvero il ripristino di parti mancanti di un segnale acustico a partire dalle parti integre del segnale. Progettiamo un sistema in grado di ripristinare intervalli multipli mancanti e di lunghezza variabile all'interno di un segnale contenente il parlato. Il nostro sistema è testato con intervalli da 100 ms fino a 1600 ms per analizzare il contributo che la visione artificiale può fornire per intervalli mancanti di durate differenti. Gli esperimenti mostrano che le prestazioni dei modelli baseline audio-only peggiorano rapidamente con l'aumentare della durata degli intervalli, mentre l'approccio audio-visivo proposto è comunque in grado di ripristinare l'informazione mancante con segnali plausibili.
Human communication is often an audio-visual experience. Indeed, listeners hear words uttered by speakers and can also see facial movements and other gestures which convey speech information. However, speech communication can be negatively affected by background noises and artifacts, which are very common in real environments. Restoring clean speech from degraded audio sources is crucial for many applications, e.g., automatic speech recognition and hearing aids. Neuroscience research proved that looking at a talking face enhances the human capability to focus auditory attention on a particular stimulus while muting external noisy sources. This dissertation is an attempt to exploit the bi-modal, i.e., audio-visual, nature of speech for speech enhancement, automatic speech recognition and speech inpainting. We start by presenting a novel approach to solve the problem of extracting the speech of a speaker of interest in a cocktail party scenario. Contrary to most previous work, we exploit a pre-trained face landmark detector and use facial landmarks motion as visual features in a deep learning model. In that way, we relieve our models from the task of learning useful visual feature from raw pixels. We train and test our models on two widely used limited size datasets and we achieve speaker independent speech enhancement in a multi-talker setting. Motivated by these results, we study how audio-visual speech enhancement can help to perform automatic speech recognition exploiting a multi-task learning framework. Then, we design a strategy where speech enhancement training phase is alternated with speech recognition phase. We observe that, in general, the joint optimization of the two phases shows a remarkable improvement of speech recognition performance compared to the audio-visual baseline models trained only to perform speech recognition. Finally, we explore if visual information can be useful for speech inpainting, i.e., the task of restoring missing parts of an acoustic speech signal from reliable audio context. We design a system that is able to inpaint multiple variable-length missing time gaps in a speech signal. We test our system with time gaps ranging from 100 ms to 1600 to investigate the contribution that vision can provide for time gaps of different duration. Experiments show that the performance of audio-only baseline models degrades rapidly when time gaps get large, while the proposed audio-visual approach is still able to plausibly restore missing information.

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22

Dookhoo, Raul. "AUTOMATED REGRESSION TESTING APPROACH TO EXPANSION AND REFINEMENT OF SPEECH RECOGNITION GRAMMARS". Master's thesis, University of Central Florida, 2008. http://digital.library.ucf.edu/cdm/ref/collection/ETD/id/2634.

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This thesis describes an approach to automated regression testing for speech recognition grammars. A prototype Audio Regression Tester called ART has been developed using Microsoft's Speech API and C#. ART allows a user to perform any of three tasks: automatically generate a new XML-based grammar file from standardized SQL database entries, record and cross-reference audio files for use by an underlying speech recognition engine, and perform regression tests with the aid of an oracle grammar. ART takes as input a wave sound file containing speech and a newly created XML grammar file. It then simultaneously executes two tests: one with the wave file and the new grammar file and the other with the wave file and the oracle grammar. The comparison result of the tests is used to determine whether the test was successful or not. This allows rapid exhaustive evaluations of additions to grammar files to guarantee forward process as the complexity of the voice domain grows. The data used in this research to derive results were taken from the LifeLike project. However, the capabilities of ART extend beyond LifeLike. The results gathered have shown that using a person's recorded voice to do regression testing is as effective as having the person do live testing. A cost-benefit analysis, using two published equations, one for Cost and the other for Benefit, was also performed to determine if automated regression testing is really more effective than manual testing. Cost captures the salaries of the engineers who perform regression testing tasks and Benefit captures revenue gains or losses related to changes in product release time. ART had a higher benefit of $21461.08 when compared to manual regression testing which had a benefit of $21393.99. Coupled with its excellent error detection rates, ART has proven to be very efficient and cost-effective in speech grammar creation and refinement.
M.S.
School of Electrical Engineering and Computer Science
Engineering and Computer Science
Computer Science MS

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23

Harvilla, Mark J. "Compensation for Nonlinear Distortion in Noise for Robust Speech Recognition". Research Showcase @ CMU, 2014. http://repository.cmu.edu/dissertations/437.

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The performance, reliability, and ubiquity of automatic speech recognition systems has flourished in recent years due to steadily increasing computational power and technological innovations such as hidden Markov models, weighted finite-state transducers, and deep learning methods. One problem which plagues speech recognition systems, especially those that operate offline and have been trained on specific in-domain data, is the deleterious effect of noise on the accuracy of speech recognition. Historically, robust speech recognition research has focused on traditional noise types such as additive noise, linear filtering, and reverberation. This thesis describes the effects of nonlinear dynamic range compression on automatic speech recognition and develops a number of novel techniques for characterizing and counteracting it. Dynamic range compression is any function which reduces the dynamic range of an input signal. Dynamic range compression is a widely-used tool in audio engineering and is almost always a component of a practical telecommunications system. Despite its ubiquity, this thesis is the first work to comprehensively study and address the effect of dynamic range compression on speech recognition. More specifically, this thesis treats the problem of dynamic range compression in three ways: (1) blind amplitude normalization methods, which counteract dynamic range compression when its parameter values allow the function to be mathematically inverted, (2) blind amplitude reconstruction techniques, i.e., declipping, which attempt to reconstruct clipped segments of the speech signal that are lost through non-invertible dynamic range compression, and (3) matched-training techniques, which attempt to select the pre-trained acoustic model with the closest set of compression parameters. All three of these methods rely on robust estimation of the dynamic range compression distortion parameters. Novel algorithms for the blind prediction of these parameters are also introduced. The algorithms' quality is evaluated in terms of the degree to which they decrease speech recognition word error rate, as well as in terms of the degree to which they increase a given speech signal's signal-to-noise ratio. In all evaluations, the possibility of independent additive noise following the application of dynamic range compression is assumed.

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24

Duckitt, William. "The design of a high-performance, floating-point embedded system for speech recognition and audio research purposes". Thesis, Link to the online version, 2008. http://hdl.handle.net/10019/824.

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25

Brady-Herbst, Brenene Marie. "An Analysis of Spondee Recognition Thresholds in Auditory-only and Audio-visual Conditions". PDXScholar, 1996. https://pdxscholar.library.pdx.edu/open_access_etds/5218.

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To date there are no acceptable speechreading tests with normative or psychometric data indicating the test is a valid and reliable measure of speechreading assessment. Middlewerd and Plomp (1987) completed a study of speechreading assessment using sentences (auditory-only and auditory-visual) in the presence of background noise. Results revealed speech reception thresholds to be lower in the auditory-visual condition. Montgomery and Demorest ( 1988) concurred that these results were appealing, but unfortunately not efficient enough to be used clinically. The purpose of this study was to develop a clinically valid and reliable assessment of speech reading ability, following Middlewerd and Plomp's ( 1987) framework to achiev~ this goal. The method of obtaining a valid assessment tool was to define a group of stimuli that can be administered and scored to produce reliable data efficiently. Because spondaic words are accepted as a reliable method of clinically achieving speech reception thresholds, they were chosen to be used as the stimuli in this study to develop an efficient clinical speechreading assessment tool. Ten subjects were presented with spondaic words in each of two conditions, auditory-only and auditory-visual, in the presence of background noise. The spondee words were randomized for each presentation, to validate the data. A computerized presentation was used so that each subject received the identical input. The computer also produced a performance-intensity function for each spondaic word. Results revealed an acceptable speech recognition threshold for 18 of the 36 spondee words in the auditory-only condition; 6 words were outside of one standard deviation; and the remaining 12 words did not produce obtainable thresholds. In the auditory-visual condition, all words except one had no obtainable threshold. Although these results invalidated the spondee words as an acceptable stimuli, the study does validate the foundation for further research to study different types of stimuli using this same framework.

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26

Choi, Hyung Keun. "Blind source separation of the audio signals in a real world". Thesis, Georgia Institute of Technology, 2002. http://hdl.handle.net/1853/14986.

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27

Zeghidour, Neil. "Learning representations of speech from the raw waveform". Thesis, Paris Sciences et Lettres (ComUE), 2019. http://www.theses.fr/2019PSLEE004/document.

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Bien que les réseaux de neurones soient à présent utilisés dans la quasi-totalité des composants d’un système de reconnaissance de la parole, du modèle acoustique au modèle de langue, l’entrée de ces systèmes reste une représentation analytique et fixée de la parole dans le domaine temps-fréquence, telle que les mel-filterbanks. Cela se distingue de la vision par ordinateur, un domaine où les réseaux de neurones prennent en entrée les pixels bruts. Les mel-filterbanks sont le produit d’une connaissance précieuse et documentée du système auditif humain, ainsi que du traitement du signal, et sont utilisées dans les systèmes de reconnaissance de la parole les plus en pointe, systèmes qui rivalisent désormais avec les humains dans certaines conditions. Cependant, les mel-filterbanks, comme toute représentation fixée, sont fondamentalement limitées par le fait qu’elles ne soient pas affinées par apprentissage pour la tâche considérée. Nous formulons l’hypothèse qu’apprendre ces représentations de bas niveau de la parole, conjontement avec le modèle, permettrait de faire avancer davantage l’état de l’art. Nous explorons tout d’abord des approches d’apprentissage faiblement supervisé et montrons que nous pouvons entraîner un unique réseau de neurones à séparer l’information phonétique de celle du locuteur à partir de descripteurs spectraux ou du signal brut et que ces représentations se transfèrent à travers les langues. De plus, apprendre à partir du signal brut produit des représentations du locuteur significativement meilleures que celles d’un modèle entraîné sur des mel-filterbanks. Ces résultats encourageants nous mènent par la suite à développer une alternative aux mel-filterbanks qui peut être entraînée à partir des données. Dans la seconde partie de cette thèse, nous proposons les Time-Domain filterbanks, une architecture neuronale légère prenant en entrée la forme d’onde, dont on peut initialiser les poids pour répliquer les mel-filterbanks et qui peut, par la suite, être entraînée par rétro-propagation avec le reste du réseau de neurones. Au cours d’expériences systématiques et approfondies, nous montrons que les Time-Domain filterbanks surclassent systématiquement les melfilterbanks, et peuvent être intégrées dans le premier système de reconnaissance de la parole purement convolutif et entraîné à partir du signal brut, qui constitue actuellement un nouvel état de l’art. Les descripteurs fixes étant également utilisés pour des tâches de classification non-linguistique, pour lesquelles elles sont d’autant moins optimales, nous entraînons un système de détection de dysarthrie à partir du signal brut, qui surclasse significativement un système équivalent entraîné sur des mel-filterbanks ou sur des descripteurs de bas niveau. Enfin, nous concluons cette thèse en expliquant en quoi nos contributions s’inscrivent dans une transition plus large vers des systèmes de compréhension du son qui pourront être appris de bout en bout
While deep neural networks are now used in almost every component of a speech recognition system, from acoustic to language modeling, the input to such systems are still fixed, handcrafted, spectral features such as mel-filterbanks. This contrasts with computer vision, in which a deep neural network is now trained on raw pixels. Mel-filterbanks contain valuable and documented prior knowledge from human auditory perception as well as signal processing, and are the input to state-of-the-art speech recognition systems that are now on par with human performance in certain conditions. However, mel-filterbanks, as any fixed representation, are inherently limited by the fact that they are not fine-tuned for the task at hand. We hypothesize that learning the low-level representation of speech with the rest of the model, rather than using fixed features, could push the state-of-the art even further. We first explore a weakly-supervised setting and show that a single neural network can learn to separate phonetic information and speaker identity from mel-filterbanks or the raw waveform, and that these representations are robust across languages. Moreover, learning from the raw waveform provides significantly better speaker embeddings than learning from mel-filterbanks. These encouraging results lead us to develop a learnable alternative to mel-filterbanks, that can be directly used in replacement of these features. In the second part of this thesis we introduce Time-Domain filterbanks, a lightweight neural network that takes the waveform as input, can be initialized as an approximation of mel-filterbanks, and then learned with the rest of the neural architecture. Across extensive and systematic experiments, we show that Time-Domain filterbanks consistently outperform melfilterbanks and can be integrated into a new state-of-the-art speech recognition system, trained directly from the raw audio signal. Fixed speech features being also used for non-linguistic classification tasks for which they are even less optimal, we perform dysarthria detection from the waveform with Time-Domain filterbanks and show that it significantly improves over mel-filterbanks or low-level descriptors. Finally, we discuss how our contributions fall within a broader shift towards fully learnable audio understanding systems

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28

Thambiratnam, Albert J. K. "Acoustic keyword spotting in speech with applications to data mining". Thesis, Queensland University of Technology, 2005. https://eprints.qut.edu.au/37254/1/Albert_Thambiratnam_Thesis.pdf.

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Keyword Spotting is the task of detecting keywords of interest within continu- ous speech. The applications of this technology range from call centre dialogue systems to covert speech surveillance devices. Keyword spotting is particularly well suited to data mining tasks such as real-time keyword monitoring and unre- stricted vocabulary audio document indexing. However, to date, many keyword spotting approaches have su®ered from poor detection rates, high false alarm rates, or slow execution times, thus reducing their commercial viability. This work investigates the application of keyword spotting to data mining tasks. The thesis makes a number of major contributions to the ¯eld of keyword spotting. The ¯rst major contribution is the development of a novel keyword veri¯cation method named Cohort Word Veri¯cation. This method combines high level lin- guistic information with cohort-based veri¯cation techniques to obtain dramatic improvements in veri¯cation performance, in particular for the problematic short duration target word class. The second major contribution is the development of a novel audio document indexing technique named Dynamic Match Lattice Spotting. This technique aug- ments lattice-based audio indexing principles with dynamic sequence matching techniques to provide robustness to erroneous lattice realisations. The resulting algorithm obtains signi¯cant improvement in detection rate over lattice-based audio document indexing while still maintaining extremely fast search speeds. The third major contribution is the study of multiple veri¯er fusion for the task of keyword veri¯cation. The reported experiments demonstrate that substantial improvements in veri¯cation performance can be obtained through the fusion of multiple keyword veri¯ers. The research focuses on combinations of speech background model based veri¯ers and cohort word veri¯ers. The ¯nal major contribution is a comprehensive study of the e®ects of limited training data for keyword spotting. This study is performed with consideration as to how these e®ects impact the immediate development and deployment of speech technologies for non-English languages.

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29

Sklar, Alexander Gabriel. "Channel Modeling Applied to Robust Automatic Speech Recognition". Scholarly Repository, 2007. http://scholarlyrepository.miami.edu/oa_theses/87.

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In automatic speech recognition systems (ASRs), training is a critical phase to the system?s success. Communication media, either analog (such as analog landline phones) or digital (VoIP) distort the speaker?s speech signal often in very complex ways: linear distortion occurs in all channels, either in the magnitude or phase spectrum. Non-linear but time-invariant distortion will always appear in all real systems. In digital systems we also have network effects which will produce packet losses and delays and repeated packets. Finally, one cannot really assert what path a signal will take, and so having error or distortion in between is almost a certainty. The channel introduces an acoustical mismatch between the speaker's signal and the trained data in the ASR, which results in poor recognition performance. The approach so far, has been to try to undo the havoc produced by the channels, i.e. compensate for the channel's behavior. In this thesis, we try to characterize the effects of different transmission media and use that as an inexpensive and repeatable way to train ASR systems.

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30

Navarathna, Rajitha Dharshana Bandara. "Robust recognition of human behaviour in challenging environments". Thesis, Queensland University of Technology, 2014. https://eprints.qut.edu.au/66235/1/Rajitha%20Dharshana%20Bandara_Navarathna_Thesis.pdf.

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Novel techniques have been developed for the automatic recognition of human behaviour in challenging environments using information from visual and infra-red camera feeds. The techniques have been applied to two interesting scenarios: Recognise drivers' speech using lip movements and recognising audience behaviour, while watching a movie, using facial features and body movements. Outcome of the research in these two areas will be useful in the improving the performance of voice recognition in automobiles for voice based control and for obtaining accurate movie interest ratings based on live audience response analysis.

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31

Martí, Guerola Amparo. "Multichannel audio processing for speaker localization, separation and enhancement". Doctoral thesis, Universitat Politècnica de València, 2013. http://hdl.handle.net/10251/33101.

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This thesis is related to the field of acoustic signal processing and its applications to emerging communication environments. Acoustic signal processing is a very wide research area covering the design of signal processing algorithms involving one or several acoustic signals to perform a given task, such as locating the sound source that originated the acquired signals, improving their signal to noise ratio, separating signals of interest from a set of interfering sources or recognizing the type of source and the content of the message. Among the above tasks, Sound Source localization (SSL) and Automatic Speech Recognition (ASR) have been specially addressed in this thesis. In fact, the localization of sound sources in a room has received a lot of attention in the last decades. Most real-word microphone array applications require the localization of one or more active sound sources in adverse environments (low signal-to-noise ratio and high reverberation). Some of these applications are teleconferencing systems, video-gaming, autonomous robots, remote surveillance, hands-free speech acquisition, etc. Indeed, performing robust sound source localization under high noise and reverberation is a very challenging task. One of the most well-known algorithms for source localization in noisy and reverberant environments is the Steered Response Power - Phase Transform (SRP-PHAT) algorithm, which constitutes the baseline framework for the contributions proposed in this thesis. Another challenge in the design of SSL algorithms is to achieve real-time performance and high localization accuracy with a reasonable number of microphones and limited computational resources. Although the SRP-PHAT algorithm has been shown to be an effective localization algorithm for real-world environments, its practical implementation is usually based on a costly fine grid-search procedure, making the computational cost of the method a real issue. In this context, several modifications and optimizations have been proposed to improve its performance and applicability. An effective strategy that extends the conventional SRP-PHAT functional is presented in this thesis. This approach performs a full exploration of the sampled space rather than computing the SRP at discrete spatial positions, increasing its robustness and allowing for a coarser spatial grid that reduces the computational cost required in a practical implementation with a small hardware cost (reduced number of microphones). This strategy allows to implement real-time applications based on location information, such as automatic camera steering or the detection of speech/non-speech fragments in advanced videoconferencing systems. As stated before, besides the contributions related to SSL, this thesis is also related to the field of ASR. This technology allows a computer or electronic device to identify the words spoken by a person so that the message can be stored or processed in a useful way. ASR is used on a day-to-day basis in a number of applications and services such as natural human-machine interfaces, dictation systems, electronic translators and automatic information desks. However, there are still some challenges to be solved. A major problem in ASR is to recognize people speaking in a room by using distant microphones. In distant-speech recognition, the microphone does not only receive the direct path signal, but also delayed replicas as a result of multi-path propagation. Moreover, there are multiple situations in teleconferencing meetings when multiple speakers talk simultaneously. In this context, when multiple speaker signals are present, Sound Source Separation (SSS) methods can be successfully employed to improve ASR performance in multi-source scenarios. This is the motivation behind the training method for multiple talk situations proposed in this thesis. This training, which is based on a robust transformed model constructed from separated speech in diverse acoustic environments, makes use of a SSS method as a speech enhancement stage that suppresses the unwanted interferences. The combination of source separation and this specific training has been explored and evaluated under different acoustical conditions, leading to improvements of up to a 35% in ASR performance.
Martí Guerola, A. (2013). Multichannel audio processing for speaker localization, separation and enhancement [Tesis doctoral no publicada]. Universitat Politècnica de València. https://doi.org/10.4995/Thesis/10251/33101
TESIS

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32

Raghunathan, Anusha. "EVALUATION OF INTELLIGIBILITY AND SPEAKER SIMILARITY OF VOICE TRANSFORMATION". UKnowledge, 2011. http://uknowledge.uky.edu/gradschool_theses/101.

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Voice transformation refers to a class of techniques that modify the voice characteristics either to conceal the identity or to mimic the voice characteristics of another speaker. Its applications include automatic dialogue replacement and voice generation for people with voice disorders. The diversity in applications makes evaluation of voice transformation a challenging task. The objective of this research is to propose a framework to evaluate intentional voice transformation techniques. Our proposed framework is based on two fundamental qualities: intelligibility and speaker similarity. Intelligibility refers to the clarity of the speech content after voice transformation and speaker similarity measures how well the modified output disguises the source speaker. We measure intelligibility with word error rates and speaker similarity with likelihood of identifying the correct speaker. The novelty of our approach is, we consider whether similarly transformed training data are available to the recognizer. We have demonstrated that this factor plays a significant role in intelligibility and speaker similarity for both human testers and automated recognizers. We thoroughly test two classes of voice transformation techniques: pitch distortion and voice conversion, using our proposed framework. We apply our results for patients with voice hypertension using video self-modeling and preliminary results are presented.

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33

Sheikh, Imran. "Exploitation du contexte sémantique pour améliorer la reconnaissance des noms propres dans les documents audio diachroniques". Thesis, Université de Lorraine, 2016. http://www.theses.fr/2016LORR0260/document.

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La nature diachronique des bulletins d'information provoque de fortes variations du contenu linguistique et du vocabulaire dans ces documents. Dans le cadre de la reconnaissance automatique de la parole, cela conduit au problème de mots hors vocabulaire (Out-Of-Vocabulary, OOV). La plupart des mots OOV sont des noms propres. Les noms propres sont très importants pour l'indexation automatique de contenus audio-vidéo. De plus, leur bonne identification est importante pour des transcriptions automatiques fiables. Le but de cette thèse est de proposer des méthodes pour récupérer les noms propres manquants dans un système de reconnaissance. Nous proposons de modéliser le contexte sémantique et d'utiliser des informations thématiques contenus dans les documents audio à transcrire. Des modèles probabilistes de thème et des projections dans un espace continu obtenues à l'aide de réseaux de neurones sont explorés pour la tâche de récupération des noms propres pertinents. Une évaluation approfondie de ces représentations contextuelles a été réalisée. Pour modéliser le contexte de nouveaux mots plus efficacement, nous proposons des réseaux de neurones qui maximisent la récupération des noms propres pertinents. En s'appuyant sur ce modèle, nous proposons un nouveau modèle (Neural Bag-of-Weighted-Words, NBOW2) qui permet d'estimer un degré d'importance pour chacun des mots du document et a la capacité de capturer des mots spécifiques à ce document. Des expériences de reconnaissance automatique de bulletins d'information télévisés montrent l'efficacité du modèle proposé. L'évaluation de NBOW2 sur d'autres tâches telles que la classification de textes montre des bonnes performances
The diachronic nature of broadcast news causes frequent variations in the linguistic content and vocabulary, leading to the problem of Out-Of-Vocabulary (OOV) words in automatic speech recognition. Most of the OOV words are found to be proper names whereas proper names are important for automatic indexing of audio-video content as well as for obtaining reliable automatic transcriptions. The goal of this thesis is to model the semantic and topical context of new proper names in order to retrieve those which are relevant to the spoken content in the audio document. Training context models is a challenging problem in this task because several new names come with a low amount of data and the context model should be robust to errors in the automatic transcription. Probabilistic topic models and word embeddings from neural network models are explored for the task of retrieval of relevant proper names. A thorough evaluation of these contextual representations is performed. It is argued that these representations, which are learned in an unsupervised manner, are not the best for the given retrieval task. Neural network context models trained with an objective to maximise the retrieval performance are proposed. The proposed Neural Bag-of-Weighted-Words (NBOW2) model learns to assign a degree of importance to input words and has the ability to capture task specific key-words. Experiments on automatic speech recognition on French broadcast news videos demonstrate the effectiveness of the proposed models. Evaluation of the NBOW2 model on standard text classification tasks shows that it learns interesting information and gives best classification accuracies among the BOW models

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34

Guenebaut, Boris. "Automatic Subtitle Generation for Sound in Videos". Thesis, University West, Department of Economics and IT, 2009. http://urn.kb.se/resolve?urn=urn:nbn:se:hv:diva-1784.

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The last ten years have been the witnesses of the emergence of any kind of video content. Moreover, the appearance of dedicated websites for this phenomenon has increased the importance the public gives to it. In the same time, certain individuals are deaf and occasionally cannot understand the meanings of such videos because there is not any text transcription available. Therefore, it is necessary to find solutions for the purpose of making these media artefacts accessible for most people. Several software propose utilities to create subtitles for videos but all require an extensive participation of the user. Thence, a more automated concept is envisaged. This thesis report indicates a way to generate subtitles following standards by using speech recognition. Three parts are distinguished. The first one consists in separating audio from video and converting the audio in suitable format if necessary. The second phase proceeds to the recognition of speech contained in the audio. The ultimate stage generates a subtitle file from the recognition results of the previous step. Directions of implementation have been proposed for the three distinct modules. The experiment results have not done enough satisfaction and adjustments have to be realized for further work. Decoding parallelization, use of well trained models, and punctuation insertion are some of the improvements to be done.

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35

Walters, Thomas C. "Auditory-based processing of communication sounds". Thesis, University of Cambridge, 2011. https://www.repository.cam.ac.uk/handle/1810/240577.

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This thesis examines the possible benefits of adapting a biologically-inspired model of human auditory processing as part of a machine-hearing system. Features were generated by an auditory model, and used as input to machine learning systems to determine the content of the sound. Features were generated using the auditory image model (AIM) and were used for speech recognition and audio search. AIM comprises processing to simulate the human cochlea, and a 'strobed temporal integration' process which generates a stabilised auditory image (SAI) from the input sound. The communication sounds which are produced by humans, other animals, and many musical instruments take the form of a pulse-resonance signal: pulses excite resonances in the body, and the resonance following each pulse contains information both about the type of object producing the sound and its size. In the case of humans, vocal tract length (VTL) determines the size properties of the resonance. In the speech recognition experiments, an auditory filterbank was combined with a Gaussian fitting procedure to produce features which are invariant to changes in speaker VTL. These features were compared against standard mel-frequency cepstral coefficients (MFCCs) in a size-invariant syllable recognition task. The VTL-invariant representation was found to produce better results than MFCCs when the system was trained on syllables from simulated talkers of one range of VTLs and tested on those from simulated talkers with a different range of VTLs. The image stabilisation process of strobed temporal integration was analysed. Based on the properties of the auditory filterbank being used, theoretical constraints were placed on the properties of the dynamic thresholding function used to perform strobe detection. These constraints were used to specify a simple, yet robust, strobe detection algorithm. The syllable recognition system described above was then extended to produce features from profiles of the SAI and tested with the same syllable database as before. For clean speech, performance of the features was comparable to that of those generated from the filterbank output. However when pink noise was added to the stimuli, performance dropped more slowly as a function of signal-to-noise ratio when using the SAI-based AIM features, than when using either the filterbank-based features or the MFCCs, demonstrating the noise-robustness properties of the SAI representation. The properties of the auditory filterbank in AIM were also analysed. Three models of the cochlea were considered: the static gammatone filterbank, dynamic compressive gammachirp (dcGC) and the pole-zero filter cascade (PZFC). The dcGC and gammatone are standard filterbank models, whereas the PZFC is a filter cascade, which more accurately models signal propagation in the cochlea. However, while the architecture of the filterbanks is different, they have all been successfully fitted to psychophysical masking data from humans. The abilities of the filterbanks to measure pitch strength were assessed, using stimuli which evoke a weak pitch percept in humans, in order to ascertain whether there is any benefit in the use of the more computationally efficient PZFC.Finally, a complete sound effects search system using auditory features was constructed in collaboration with Google research. Features were computed from the SAI by sampling the SAI space with boxes of different scales. Vector quantization (VQ) was used to convert this multi-scale representation to a sparse code. The 'passive-aggressive model for image retrieval' (PAMIR) was used to learn the relationships between dictionary words and these auditory codewords. These auditory sparse codes were compared against sparse codes generated from MFCCs, and the best performance was found when using the auditory features.

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Fong, Katherine KaYan. "IR-Depth Face Detection and Lip Localization Using Kinect V2". DigitalCommons@CalPoly, 2015. https://digitalcommons.calpoly.edu/theses/1425.

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Face recognition and lip localization are two main building blocks in the development of audio visual automatic speech recognition systems (AV-ASR). In many earlier works, face recognition and lip localization were conducted in uniform lighting conditions with simple backgrounds. However, such conditions are seldom the case in real world applications. In this paper, we present an approach to face recognition and lip localization that is invariant to lighting conditions. This is done by employing infrared and depth images captured by the Kinect V2 device. First we present the use of infrared images for face detection. Second, we use the face’s inherent depth information to reduce the search area for the lips by developing a nose point detection. Third, we further reduce the search area by using a depth segmentation algorithm to separate the face from its background. Finally, with the reduced search range, we present a method for lip localization based on depth gradients. Experimental results demonstrated an accuracy of 100% for face detection, and 96% for lip localization.

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Wallace, Roy Geoffrey. "Fast and accurate phonetic spoken term detection". Thesis, Queensland University of Technology, 2010. https://eprints.qut.edu.au/39610/1/Roy_Wallace_Thesis.pdf.

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For the first time in human history, large volumes of spoken audio are being broadcast, made available on the internet, archived, and monitored for surveillance every day. New technologies are urgently required to unlock these vast and powerful stores of information. Spoken Term Detection (STD) systems provide access to speech collections by detecting individual occurrences of specified search terms. The aim of this work is to develop improved STD solutions based on phonetic indexing. In particular, this work aims to develop phonetic STD systems for applications that require open-vocabulary search, fast indexing and search speeds, and accurate term detection. Within this scope, novel contributions are made within two research themes, that is, accommodating phone recognition errors and, secondly, modelling uncertainty with probabilistic scores. A state-of-the-art Dynamic Match Lattice Spotting (DMLS) system is used to address the problem of accommodating phone recognition errors with approximate phone sequence matching. Extensive experimentation on the use of DMLS is carried out and a number of novel enhancements are developed that provide for faster indexing, faster search, and improved accuracy. Firstly, a novel comparison of methods for deriving a phone error cost model is presented to improve STD accuracy, resulting in up to a 33% improvement in the Figure of Merit. A method is also presented for drastically increasing the speed of DMLS search by at least an order of magnitude with no loss in search accuracy. An investigation is then presented of the effects of increasing indexing speed for DMLS, by using simpler modelling during phone decoding, with results highlighting the trade-off between indexing speed, search speed and search accuracy. The Figure of Merit is further improved by up to 25% using a novel proposal to utilise word-level language modelling during DMLS indexing. Analysis shows that this use of language modelling can, however, be unhelpful or even disadvantageous for terms with a very low language model probability. The DMLS approach to STD involves generating an index of phone sequences using phone recognition. An alternative approach to phonetic STD is also investigated that instead indexes probabilistic acoustic scores in the form of a posterior-feature matrix. A state-of-the-art system is described and its use for STD is explored through several experiments on spontaneous conversational telephone speech. A novel technique and framework is proposed for discriminatively training such a system to directly maximise the Figure of Merit. This results in a 13% improvement in the Figure of Merit on held-out data. The framework is also found to be particularly useful for index compression in conjunction with the proposed optimisation technique, providing for a substantial index compression factor in addition to an overall gain in the Figure of Merit. These contributions significantly advance the state-of-the-art in phonetic STD, by improving the utility of such systems in a wide range of applications.

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Zhezhela, Oleksandr. "Vizualizace výstupu z řečových technologií pro potřeby kontaktních center". Master's thesis, Vysoké učení technické v Brně. Fakulta informačních technologií, 2014. http://www.nusl.cz/ntk/nusl-236041.

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The thesis is aimed on visualisation of data mined by speech processing technologies. Some methods speech data extraction were studied and technologies for this task were analysed. The variety of meta data that can be mined from speech was defined. Were also examined existing standards and processes of call centres. Some requirements for the user interface were gathered and analysed. On that basis and after communication with call centre employees there was defined and implemented a concept for speech data visualization. Gained solutions were integrated into Speech Analytics Server (SPAS).

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Kalantari, Shahram. "Improving spoken term detection using complementary information". Thesis, Queensland University of Technology, 2015. https://eprints.qut.edu.au/90074/1/Shahram_Kalantari_Thesis.pdf.

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This research has made contributions to the area of spoken term detection (STD), defined as the process of finding all occurrences of a specified search term in a large collection of speech segments. The use of visual information in the form of lip movements of the speaker in addition to audio and the use of topic of the speech segments, and the expected frequency of words in the target speech domain, are proposed. By using these complementary information, improvement in the performance of STD has been achieved which enables efficient search of key words in large collection of multimedia documents.

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40

Lucey, Patrick Joseph. "Lipreading across multiple views". Thesis, Queensland University of Technology, 2007. https://eprints.qut.edu.au/16676/1/Patrick_Joseph_Lucey_Thesis.pdf.

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Visual information from a speaker's mouth region is known to improve automatic speech recognition (ASR) robustness, especially in the presence of acoustic noise. Currently, the vast majority of audio-visual ASR (AVASR) studies assume frontal images of the speaker's face, which is a rather restrictive human-computer interaction (HCI) scenario. The lack of research into AVASR across multiple views has been dictated by the lack of large corpora that contains varying pose/viewpoint speech data. Recently, research has concentrated on recognising human be- haviours within "meeting " or "lecture " type scenarios via "smart-rooms ". This has resulted in the collection of audio-visual speech data which allows for the recognition of visual speech from both frontal and non-frontal views to occur. Using this data, the main focus of this thesis was to investigate and develop various methods within the confines of a lipreading system which can recognise visual speech across multiple views. This reseach constitutes the first published work within the field which looks at this particular aspect of AVASR. The task of recognising visual speech from non-frontal views (i.e. profile) is in principle very similar to that of frontal views, requiring the lipreading system to initially locate and track the mouth region and subsequently extract visual features. However, this task is far more complicated than the frontal case, because the facial features required to locate and track the mouth lie in a much more limited spatial plane. Nevertheless, accurate mouth region tracking can be achieved by employing techniques similar to frontal facial feature localisation. Once the mouth region has been extracted, the same visual feature extraction process can take place to the frontal view. A novel contribution of this thesis, is to quantify the degradation in lipreading performance between the frontal and profile views. In addition to this, novel patch-based analysis of the various views is conducted, and as a result a novel multi-stream patch-based representation is formulated. Having a lipreading system which can recognise visual speech from both frontal and profile views is a novel contribution to the field of AVASR. How- ever, given both the frontal and profile viewpoints, this begs the question, is there any benefit of having the additional viewpoint? Another major contribution of this thesis, is an exploration of a novel multi-view lipreading system. This system shows that there does exist complimentary information in the additional viewpoint (possibly that of lip protrusion), with superior performance achieved in the multi-view system compared to the frontal-only system. Even though having a multi-view lipreading system which can recognise visual speech from both front and profile views is very beneficial, it can hardly considered to be realistic, as each particular viewpoint is dedicated to a single pose (i.e. front or profile). In an effort to make the lipreading system more realistic, a unified system based on a single camera was developed which enables a lipreading system to recognise visual speech from both frontal and profile poses. This is called pose-invariant lipreading. Pose-invariant lipreading can be performed on either stationary or continuous tasks. Methods which effectively normalise the various poses into a single pose were investigated for the stationary scenario and in another contribution of this thesis, an algorithm based on regularised linear regression was employed to project all the visual speech features into a uniform pose. This particular method is shown to be beneficial when the lipreading system was biased towards the dominant pose (i.e. frontal). The final contribution of this thesis is the formulation of a continuous pose-invariant lipreading system which contains a pose-estimator at the start of the visual front-end. This system highlights the complexity of developing such a system, as introducing more flexibility within the lipreading system invariability means the introduction of more error. All the works contained in this thesis present novel and innovative contributions to the field of AVASR, and hopefully this will aid in the future deployment of an AVASR system in realistic scenarios.

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Lucey, Patrick Joseph. "Lipreading across multiple views". Queensland University of Technology, 2007. http://eprints.qut.edu.au/16676/.

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Visual information from a speaker's mouth region is known to improve automatic speech recognition (ASR) robustness, especially in the presence of acoustic noise. Currently, the vast majority of audio-visual ASR (AVASR) studies assume frontal images of the speaker's face, which is a rather restrictive human-computer interaction (HCI) scenario. The lack of research into AVASR across multiple views has been dictated by the lack of large corpora that contains varying pose/viewpoint speech data. Recently, research has concentrated on recognising human be- haviours within "meeting " or "lecture " type scenarios via "smart-rooms ". This has resulted in the collection of audio-visual speech data which allows for the recognition of visual speech from both frontal and non-frontal views to occur. Using this data, the main focus of this thesis was to investigate and develop various methods within the confines of a lipreading system which can recognise visual speech across multiple views. This reseach constitutes the first published work within the field which looks at this particular aspect of AVASR. The task of recognising visual speech from non-frontal views (i.e. profile) is in principle very similar to that of frontal views, requiring the lipreading system to initially locate and track the mouth region and subsequently extract visual features. However, this task is far more complicated than the frontal case, because the facial features required to locate and track the mouth lie in a much more limited spatial plane. Nevertheless, accurate mouth region tracking can be achieved by employing techniques similar to frontal facial feature localisation. Once the mouth region has been extracted, the same visual feature extraction process can take place to the frontal view. A novel contribution of this thesis, is to quantify the degradation in lipreading performance between the frontal and profile views. In addition to this, novel patch-based analysis of the various views is conducted, and as a result a novel multi-stream patch-based representation is formulated. Having a lipreading system which can recognise visual speech from both frontal and profile views is a novel contribution to the field of AVASR. How- ever, given both the frontal and profile viewpoints, this begs the question, is there any benefit of having the additional viewpoint? Another major contribution of this thesis, is an exploration of a novel multi-view lipreading system. This system shows that there does exist complimentary information in the additional viewpoint (possibly that of lip protrusion), with superior performance achieved in the multi-view system compared to the frontal-only system. Even though having a multi-view lipreading system which can recognise visual speech from both front and profile views is very beneficial, it can hardly considered to be realistic, as each particular viewpoint is dedicated to a single pose (i.e. front or profile). In an effort to make the lipreading system more realistic, a unified system based on a single camera was developed which enables a lipreading system to recognise visual speech from both frontal and profile poses. This is called pose-invariant lipreading. Pose-invariant lipreading can be performed on either stationary or continuous tasks. Methods which effectively normalise the various poses into a single pose were investigated for the stationary scenario and in another contribution of this thesis, an algorithm based on regularised linear regression was employed to project all the visual speech features into a uniform pose. This particular method is shown to be beneficial when the lipreading system was biased towards the dominant pose (i.e. frontal). The final contribution of this thesis is the formulation of a continuous pose-invariant lipreading system which contains a pose-estimator at the start of the visual front-end. This system highlights the complexity of developing such a system, as introducing more flexibility within the lipreading system invariability means the introduction of more error. All the works contained in this thesis present novel and innovative contributions to the field of AVASR, and hopefully this will aid in the future deployment of an AVASR system in realistic scenarios.

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Temko, Andriy. "Acoustic event detection and classification". Doctoral thesis, Universitat Politècnica de Catalunya, 2007. http://hdl.handle.net/10803/6880.

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L'activitat humana que té lloc en sales de reunions o aules d'ensenyament es veu reflectida en una rica varietat d'events acústics, ja siguin produïts pel cos humà o per objectes que les persones manegen. Per això, la determinació de la identitat dels sons i de la seva posició temporal pot ajudar a detectar i a descriure l'activitat humana que té lloc en la sala. A més a més, la detecció de sons diferents de la veu pot ajudar a millorar la robustes de tecnologies de la parla com el reconeixement automàtica a condicions de treball adverses. L'objectiu d'aquesta tesi és la detecció i classificació automàtica d'events acústics. Es tracta de processar els senyals acústics recollits per micròfons distants en sales de reunions o aules per tal de convertir-los en descripcions simbòliques que es corresponguin amb la percepció que un oient tindria dels diversos events sonors continguts en els senyals i de les seves fonts. En primer lloc, s'encara la tasca de classificació automàtica d'events acústics amb classificadors de màquines de vectors suport (Support Vector Machines (SVM)), elecció motivada per l'escassetat de dades d'entrenament. Per al problema de reconeixement multiclasse es desenvolupa un esquema d'agrupament automàtic amb conjunt de característiques variable i basat en matrius de confusió. Realitzant proves amb la base de dades recollida, aquest classificador obté uns millors resultats que la tècnica basada en models de barreges de Gaussianes (Gaussian Mixture Models (GMM)), i aconsegueix una reducció relativa de l'error mitjà elevada en comparació amb el millor resultat obtingut amb l'esquema convencional basat en arbre binari. Continuant amb el problema de classificació, es comparen unes quantes maneres alternatives d'estendre els SVM al processament de seqüències, en un intent d'evitar l'inconvenient de treballar amb vectors de longitud fixa que presenten els SVM quan han de tractar dades d'àudio. En aquestes proves s'observa que els nuclis de deformació temporal dinàmica funcionen bé amb sons que presenten una estructura temporal. A més a més, s'usen conceptes i eines manllevats de la teoria de lògica difusa per investigar, d'una banda, la importància de cada una de les característiques i el grau d'interacció entre elles, i d'altra banda, tot cercant l'augment de la taxa de classificació, s'investiga la fusió de les
sortides de diversos sistemes de classificació. Els sistemes de classificació d'events acústics
desenvolupats s'han testejat també mitjançant la participació en unes quantes avaluacions d'àmbit
internacional, entre els anys 2004 i 2006. La segona principal contribució d'aquest treball de tesi consisteix en el desenvolupament de sistemes de detecció d'events acústics. El problema de la detecció és més complex, ja que inclou tant la classificació dels sons com la determinació dels intervals temporals on tenen lloc. Es desenvolupen dues versions del sistema i es proven amb els conjunts de dades de les dues campanyes d'avaluació internacional CLEAR que van tenir lloc els anys 2006 i 2007, fent-se servir dos tipus de bases de dades: dues bases d'events acústics aïllats, i una base d'enregistraments de seminaris interactius, les quals contenen un nombre relativament elevat d'ocurrències dels events acústics especificats. Els sistemes desenvolupats, que consisteixen en l'ús de classificadors basats en SVM que operen dins
d'una finestra lliscant més un post-processament, van ser els únics presentats a les avaluacions
esmentades que no es basaven en models de Markov ocults (Hidden Markov Models) i cada un d'ells
va obtenir resultats competitius en la corresponent avaluació. La detecció d'activitat oral és un altre dels objectius d'aquest treball de tesi, pel fet de ser un cas particular de detecció d'events acústics especialment important. Es desenvolupa una tècnica de millora de l'entrenament dels SVM per fer front a la necessitat de reducció de l'enorme conjunt de dades existents. El sistema resultant, basat en SVM, és testejat amb uns quants conjunts de dades de l'avaluació NIST RT (Rich Transcription), on mostra puntuacions millors que les del sistema basat en GMM, malgrat que aquest darrer va quedar entre els primers en l'avaluació NIST RT de 2006.
Per acabar, val la pena esmentar alguns resultats col·laterals d'aquest treball de tesi. Com que s'ha dut a terme en l'entorn del projecte europeu CHIL, l'autor ha estat responsable de l'organització de les avaluacions internacionals de classificació i detecció d'events acústics abans esmentades, liderant l'especificació de les classes d'events, les bases de dades, els protocols d'avaluació i, especialment, proposant i implementant les diverses mètriques utilitzades. A més a més, els sistemes de detecció
s'han implementat en la sala intel·ligent de la UPC, on funcionen en temps real a efectes de test i demostració.
The human activity that takes place in meeting-rooms or class-rooms is reflected in a rich variety of acoustic events, either produced by the human body or by objects handled by humans, so the determination of both the identity of sounds and their position in time may help to detect and describe that human activity.
Additionally, detection of sounds other than speech may be useful to enhance the robustness of speech technologies like automatic speech recognition. Automatic detection and classification of acoustic events is the objective of this thesis work. It aims at processing the acoustic signals collected by distant microphones in meeting-room or classroom environments to convert them into symbolic descriptions corresponding to a listener's perception of the different sound events that are present in the signals and their sources. First of all, the task of acoustic event classification is faced using Support Vector Machine (SVM) classifiers, which are motivated by the scarcity of training data. A confusion-matrix-based variable-feature-set clustering scheme is developed for the multiclass recognition problem, and tested on the gathered database. With it, a higher classification rate than the GMM-based technique is obtained, arriving to a large relative average error reduction with respect to the best result from the conventional binary tree scheme. Moreover, several ways to extend SVMs to sequence processing are compared, in an attempt to avoid the drawback of SVMs when dealing with audio data, i.e. their restriction to work with fixed-length vectors, observing that the dynamic time warping kernels work well for sounds that show a temporal structure. Furthermore, concepts and tools from the fuzzy theory are used to investigate, first, the importance of and degree of interaction among features, and second, ways to fuse the outputs of several classification systems. The developed AEC systems are tested also by participating in several international evaluations from 2004 to 2006, and the results
are reported. The second main contribution of this thesis work is the development of systems for detection of acoustic events. The detection problem is more complex since it includes both classification and determination of the time intervals where the sound takes place. Two system versions are developed and tested on the datasets of the two CLEAR international evaluation campaigns in 2006 and 2007. Two kinds of databases are used: two databases of isolated acoustic events, and a database of interactive seminars containing a significant number of acoustic events of interest. Our developed systems, which consist of SVM-based classification within a sliding window plus post-processing, were the only submissions not using HMMs, and each of them obtained competitive results in the corresponding evaluation. Speech activity detection was also pursued in this thesis since, in fact, it is a -especially important - particular case of acoustic event detection. An enhanced SVM training approach for the speech activity detection task is developed, mainly to cope with the problem of dataset reduction. The resulting SVM-based system is tested with several NIST Rich Transcription (RT) evaluation datasets, and it shows better scores than our GMM-based system, which ranked among the best systems in the RT06 evaluation. Finally, it is worth mentioning a few side outcomes from this thesis work. As it has been carried out in the framework of the CHIL EU project, the author has been responsible for the organization of the above mentioned international evaluations in acoustic event classification and detection, taking a leading role in the specification of acoustic event classes, databases, and evaluation protocols, and, especially, in the proposal and implementation of the various metrics that have been used. Moreover, the detection systems have been implemented in the UPC's smart-room and work in real time for purposes of testing and demonstration.

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Verdet, Florian. "Exploring variabilities through factor analysis in automatic acoustic language recognition". Phd thesis, Université d'Avignon, 2011. http://tel.archives-ouvertes.fr/tel-00954255.

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Language Recognition is the problem of discovering the language of a spoken definitionutterance. This thesis achieves this goal by using short term acoustic information within a GMM-UBM approach.The main problem of many pattern recognition applications is the variability of problemthe observed data. In the context of Language Recognition (LR), this troublesomevariability is due to the speaker characteristics, speech evolution, acquisition and transmission channels.In the context of Speaker Recognition, the variability problem is solved by solutionthe Joint Factor Analysis (JFA) technique. Here, we introduce this paradigm toLanguage Recognition. The success of JFA relies on several assumptions: The globalJFA assumption is that the observed information can be decomposed into a universalglobal part, a language-dependent part and the language-independent variabilitypart. The second, more technical assumption consists in the unwanted variability part to be thought to live in a low-dimensional, globally defined subspace. In this work, we analyze how JFA behaves in the context of a GMM-UBM LR framework. We also introduce and analyze its combination with Support Vector Machines(SVMs).The first JFA publications put all unwanted information (hence the variability) improvemen tinto one and the same component, which is thought to follow a Gaussian distribution.This handles diverse kinds of variability in a unique manner. But in practice,we observe that this hypothesis is not always verified. We have for example thecase, where the data can be divided into two clearly separate subsets, namely datafrom telephony and from broadcast sources. In this case, our detailed investigations show that there is some benefit of handling the two kinds of data with two separatesystems and then to elect the output score of the system, which corresponds to the source of the testing utterance.For selecting the score of one or the other system, we need a channel source related analyses detector. We propose here different novel designs for such automatic detectors.In this framework, we show that JFA's variability factors (of the subspace) can beused with success for detecting the source. This opens the interesting perspectiveof partitioning the data into automatically determined channel source categories,avoiding the need of source-labeled training data, which is not always available.The JFA approach results in up to 72% relative cost reduction, compared to the overall resultsGMM-UBM baseline system. Using source specific systems followed by a scoreselector, we achieve 81% relative improvement.

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Silvestre, Cerdà Joan Albert. "Different Contributions to Cost-Effective Transcription and Translation of Video Lectures". Doctoral thesis, Universitat Politècnica de València, 2016. http://hdl.handle.net/10251/62194.

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[EN] In recent years, on-line multimedia repositories have experiencied a strong growth that have made them consolidated as essential knowledge assets, especially in the area of education, where large repositories of video lectures have been built in order to complement or even replace traditional teaching methods. However, most of these video lectures are neither transcribed nor translated due to a lack of cost-effective solutions to do so in a way that gives accurate enough results. Solutions of this kind are clearly necessary in order to make these lectures accessible to speakers of different languages and to people with hearing disabilities. They would also facilitate lecture searchability and analysis functions, such as classification, recommendation or plagiarism detection, as well as the development of advanced educational functionalities like content summarisation to assist student note-taking. For this reason, the main aim of this thesis is to develop a cost-effective solution capable of transcribing and translating video lectures to a reasonable degree of accuracy. More specifically, we address the integration of state-of-the-art techniques in Automatic Speech Recognition and Machine Translation into large video lecture repositories to generate high-quality multilingual video subtitles without human intervention and at a reduced computational cost. Also, we explore the potential benefits of the exploitation of the information that we know a priori about these repositories, that is, lecture-specific knowledge such as speaker, topic or slides, to create specialised, in-domain transcription and translation systems by means of massive adaptation techniques. The proposed solutions have been tested in real-life scenarios by carrying out several objective and subjective evaluations, obtaining very positive results. The main outcome derived from this thesis, The transLectures-UPV Platform, has been publicly released as an open-source software, and, at the time of writing, it is serving automatic transcriptions and translations for several thousands of video lectures in many Spanish and European universities and institutions.
[ES] Durante estos últimos años, los repositorios multimedia on-line han experimentado un gran crecimiento que les ha hecho establecerse como fuentes fundamentales de conocimiento, especialmente en el área de la educación, donde se han creado grandes repositorios de vídeo charlas educativas para complementar e incluso reemplazar los métodos de enseñanza tradicionales. No obstante, la mayoría de estas charlas no están transcritas ni traducidas debido a la ausencia de soluciones de bajo coste que sean capaces de hacerlo garantizando una calidad mínima aceptable. Soluciones de este tipo son claramente necesarias para hacer que las vídeo charlas sean más accesibles para hablantes de otras lenguas o para personas con discapacidades auditivas. Además, dichas soluciones podrían facilitar la aplicación de funciones de búsqueda y de análisis tales como clasificación, recomendación o detección de plagios, así como el desarrollo de funcionalidades educativas avanzadas, como por ejemplo la generación de resúmenes automáticos de contenidos para ayudar al estudiante a tomar apuntes. Por este motivo, el principal objetivo de esta tesis es desarrollar una solución de bajo coste capaz de transcribir y traducir vídeo charlas con un nivel de calidad razonable. Más específicamente, abordamos la integración de técnicas estado del arte de Reconocimiento del Habla Automático y Traducción Automática en grandes repositorios de vídeo charlas educativas para la generación de subtítulos multilingües de alta calidad sin requerir intervención humana y con un reducido coste computacional. Además, también exploramos los beneficios potenciales que conllevaría la explotación de la información de la que disponemos a priori sobre estos repositorios, es decir, conocimientos específicos sobre las charlas tales como el locutor, la temática o las transparencias, para crear sistemas de transcripción y traducción especializados mediante técnicas de adaptación masiva. Las soluciones propuestas en esta tesis han sido testeadas en escenarios reales llevando a cabo nombrosas evaluaciones objetivas y subjetivas, obteniendo muy buenos resultados. El principal legado de esta tesis, The transLectures-UPV Platform, ha sido liberado públicamente como software de código abierto, y, en el momento de escribir estas líneas, está sirviendo transcripciones y traducciones automáticas para diversos miles de vídeo charlas educativas en nombrosas universidades e instituciones Españolas y Europeas.
[CAT] Durant aquests darrers anys, els repositoris multimèdia on-line han experimentat un gran creixement que els ha fet consolidar-se com a fonts fonamentals de coneixement, especialment a l'àrea de l'educació, on s'han creat grans repositoris de vídeo xarrades educatives per tal de complementar o inclús reemplaçar els mètodes d'ensenyament tradicionals. No obstant això, la majoria d'aquestes xarrades no estan transcrites ni traduïdes degut a l'absència de solucions de baix cost capaces de fer-ho garantint una qualitat mínima acceptable. Solucions d'aquest tipus són clarament necessàries per a fer que les vídeo xarres siguen més accessibles per a parlants d'altres llengües o per a persones amb discapacitats auditives. A més, aquestes solucions podrien facilitar l'aplicació de funcions de cerca i d'anàlisi tals com classificació, recomanació o detecció de plagis, així com el desenvolupament de funcionalitats educatives avançades, com per exemple la generació de resums automàtics de continguts per ajudar a l'estudiant a prendre anotacions. Per aquest motiu, el principal objectiu d'aquesta tesi és desenvolupar una solució de baix cost capaç de transcriure i traduir vídeo xarrades amb un nivell de qualitat raonable. Més específicament, abordem la integració de tècniques estat de l'art de Reconeixement de la Parla Automàtic i Traducció Automàtica en grans repositoris de vídeo xarrades educatives per a la generació de subtítols multilingües d'alta qualitat sense requerir intervenció humana i amb un reduït cost computacional. A més, també explorem els beneficis potencials que comportaria l'explotació de la informació de la que disposem a priori sobre aquests repositoris, és a dir, coneixements específics sobre les xarrades tals com el locutor, la temàtica o les transparències, per a crear sistemes de transcripció i traducció especialitzats mitjançant tècniques d'adaptació massiva. Les solucions proposades en aquesta tesi han estat testejades en escenaris reals duent a terme nombroses avaluacions objectives i subjectives, obtenint molt bons resultats. El principal llegat d'aquesta tesi, The transLectures-UPV Platform, ha sigut alliberat públicament com a programari de codi obert, i, en el moment d'escriure aquestes línies, està servint transcripcions i traduccions automàtiques per a diversos milers de vídeo xarrades educatives en nombroses universitats i institucions Espanyoles i Europees.
Silvestre Cerdà, JA. (2016). Different Contributions to Cost-Effective Transcription and Translation of Video Lectures [Tesis doctoral no publicada]. Universitat Politècnica de València. https://doi.org/10.4995/Thesis/10251/62194
TESIS

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45

Fernández, López Adriana. "Learning of meaningful visual representations for continuous lip-reading". Doctoral thesis, Universitat Pompeu Fabra, 2021. http://hdl.handle.net/10803/671206.

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In the last decades, there has been an increased interest in decoding speech exclusively using visual cues, i.e. mimicking the human capability to perform lip-reading, leading to Automatic Lip-Reading (ALR) systems. However, it is well known that the access to speech through the visual channel is subject to many limitations when compared to the audio channel, i.e. it has been argued that humans can actually read around 30% of the information from the lips, and the rest is filled-in from the context. Thus, one of the main challenges in ALR resides in the visual ambiguities that arise at the word level, highlighting that not all sounds that we hear can be easily distinguished by observing the lips. In the literature, early ALR systems addressed simple recognition tasks such as alphabet or digit recognition but progressively shifted to more complex and realistic settings leading to several recent systems that target continuous lip-reading. To a large extent, these advances have been possible thanks to the construction of powerful systems based on deep learning architectures that have quickly started to replace traditional systems. Despite the recognition rates for continuous lip-reading may appear modest in comparison to those achieved by audio-based systems, the field has undeniably made a step forward. Interestingly, an analogous effect can be observed when humans try to decode speech: given sufficiently clean signals, most people can effortlessly decode the audio channel but would struggle to perform lip-reading, since the ambiguity of the visual cues makes it necessary the use of further context to decode the message. In this thesis, we explore the appropriate modeling of visual representations with the aim to improve continuous lip-reading. To this end, we present different data-driven mechanisms to handle the main challenges in lip-reading related to the ambiguities or the speaker dependency of visual cues. Our results highlight the benefits of a proper encoding of the visual channel, for which the most useful features are those that encode corresponding lip positions in a similar way, independently of the speaker. This fact opens the door to i) lip-reading in many different languages without requiring large-scale datasets, and ii) increasing the contribution of the visual channel in audio-visual speech systems. On the other hand, our experiments identify a tendency to focus on the modeling of temporal context as the key to advance the field, where there is a need for ALR models that are trained on datasets comprising large speech variability at several context levels. In this thesis, we show that both proper modeling of visual representations and the ability to retain context at several levels are necessary conditions to build successful lip-reading systems.
En les darreres dècades, hi ha hagut un interès creixent en la descodificació de la parla utilitzant exclusivament senyals visuals, es a dir, imitant la capacitat humana de llegir els llavis, donant lloc a sistemes de lectura automàtica de llavis (ALR). No obstant això, se sap que l’accès a la parla a través del canal visual està subjecte a moltes limitacions en comparació amb el senyal acústic, es a dir, s’ha argumentat que els humans poden llegir al voltant del 30% de la informació dels llavis, i la resta es completa fent servir el context. Així, un dels principals reptes de l’ALR resideix en les ambigüitats visuals que sorgeixen a escala de paraula, destacant que no tots els sons que escoltem es poden distingir fàcilment observant els llavis. A la literatura, els primers sistemes ALR van abordar tasques de reconeixement senzilles, com ara el reconeixement de l’alfabet o els dígits, però progressivament van passar a entorns mes complexos i realistes que han conduït a diversos sistemes recents dirigits a la lectura continua dels llavis. En gran manera, aquests avenços han estat possibles gracies a la construcció de sistemes potents basats en arquitectures d’aprenentatge profund que han començat a substituir ràpidament els sistemes tradicionals. Tot i que les taxes de reconeixement de la lectura continua dels llavis poden semblar modestes en comparació amb les assolides pels sistemes basats en audio, és evident que el camp ha fet un pas endavant. Curiosament, es pot observar un efecte anàleg quan els humans intenten descodificar la parla: donats senyals sense soroll, la majoria de la gent pot descodificar el canal d’àudio sense esforç¸, però tindria dificultats per llegir els llavis, ja que l’ambigüitat dels senyals visuals fa necessari l’ús de context addicional per descodificar el missatge. En aquesta tesi explorem el modelatge adequat de representacions visuals amb l’objectiu de millorar la lectura contínua dels llavis. Amb aquest objectiu, presentem diferents mecanismes basats en dades per fer front als principals reptes de la lectura de llavis relacionats amb les ambigüitats o la dependència dels parlants dels senyals visuals. Els nostres resultats destaquen els avantatges d’una correcta codificació del canal visual, per a la qual les característiques més útils són aquelles que codifiquen les posicions corresponents dels llavis d’una manera similar, independentment de l’orador. Aquest fet obre la porta a i) la lectura de llavis en molts idiomes diferents sense necessitat de conjunts de dades a gran escala, i ii) a l’augment de la contribució del canal visual en sistemes de parla audiovisuals.´ D’altra banda, els nostres experiments identifiquen una tendència a centrar-se en iii la modelització del context temporal com la clau per avançar en el camp, on hi ha la necessitat de models d’ALR que s’entrenin en conjunts de dades que incloguin una gran variabilitat de la parla a diversos nivells de context. En aquesta tesi, demostrem que tant el modelatge adequat de les representacions visuals com la capacitat de retenir el context a diversos nivells són condicions necessàries per construir sistemes de lectura de llavis amb èxit.

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46

Jain, Abhilash. "Visual Speech Recognition". Thesis, 2018. https://etd.iisc.ac.in/handle/2005/4767.

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Visual speech recognition (VSR), or automatic lip-reading, is the task of extracting speech information from visual input. The addition of visual speech has been shown to improve the performance of traditional audio speech recognition (ASR) systems, and hence has been active area of research since it's inception. This thesis proposes a new VSR system for isolated word recognition tasks, with focus on the feature extraction methodology. A novel two-stage feature extraction technique is proposed. Image transform based features { discrete cosine transform (DCT) and local binary patterns (LBP) { are used. The use of di erence images for temporal feature extraction is also proposed. A new region of interest (ROI), which consists of the throat and lower jaw along with the mouth, is also introduced. For ROI extraction, the Viola-Jones algorithm is used. Classi cation is done using a multi-class Support Vector Machine (SVM) model. The system provides a simple, yet effective way to extract features from the video input, and performs comparably to some recent VSR systems, which employ more complicated techniques, like lip modelling or deep learning, to extract visual features.

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47

Liu, Feng. "Audio fingerprinting for speech reconstruction and recognition in noisy environments". Thesis, 2017. http://hdl.handle.net/1828/7912.

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Audio fingerprinting is a highly specific content-based audio retrieval technique. Given a short audio fragment as query, an audio fingerprinting system can identify the particular file that contains the fragment in a large library potentially consisting of millions of audio files. In this thesis, we investigate the possibility and feasibility of applying audio fingerprinting to do speech recognition in noisy environments based on speech reconstruction. To reconstruct noisy speech, the speech is divided into small segments of equal length at first. Then, audio fingerprinting is used to find the most similar segment in a large dataset consisting of clean speech files. If the similarity is above a threshold, the noisy segment is replaced with the clean segment. At last, all the segments, after conditional replacement, are concatenated to form the reconstructed speech, which is sent to a traditional speech recognition system. In the above procedure, a critical step is using audio fingerprinting to find the clean speech segment in a dataset. To test its performance, we build a landmark-based audio fingerprinting system. Experimental results show that this baseline system performs well in traditional applications, but its accuracy in this new application is not as good as we expected. Next, we propose three strategies to improve the system, resulting in better accuracy than the baseline system. Finally, we integrate the improved audio fingerprinting system into a traditional speech recognition system and evaluate the performance of the whole system.
Graduate

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48

Makkook, Mustapha. "A Multimodal Sensor Fusion Architecture for Audio-Visual Speech Recognition". Thesis, 2007. http://hdl.handle.net/10012/3065.

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A key requirement for developing any innovative system in a computing environment is to integrate a sufficiently friendly interface with the average end user. Accurate design of such a user-centered interface, however, means more than just the ergonomics of the panels and displays. It also requires that designers precisely define what information to use and how, where, and when to use it. Recent advances in user-centered design of computing systems have suggested that multimodal integration can provide different types and levels of intelligence to the user interface. The work of this thesis aims at improving speech recognition-based interfaces by making use of the visual modality conveyed by the movements of the lips. Designing a good visual front end is a major part of this framework. For this purpose, this work derives the optical flow fields for consecutive frames of people speaking. Independent Component Analysis (ICA) is then used to derive basis flow fields. The coefficients of these basis fields comprise the visual features of interest. It is shown that using ICA on optical flow fields yields better classification results than the traditional approaches based on Principal Component Analysis (PCA). In fact, ICA can capture higher order statistics that are needed to understand the motion of the mouth. This is due to the fact that lips movement is complex in its nature, as it involves large image velocities, self occlusion (due to the appearance and disappearance of the teeth) and a lot of non-rigidity. Another issue that is of great interest to audio-visual speech recognition systems designers is the integration (fusion) of the audio and visual information into an automatic speech recognizer. For this purpose, a reliability-driven sensor fusion scheme is developed. A statistical approach is developed to account for the dynamic changes in reliability. This is done in two steps. The first step derives suitable statistical reliability measures for the individual information streams. These measures are based on the dispersion of the N-best hypotheses of the individual stream classifiers. The second step finds an optimal mapping between the reliability measures and the stream weights that maximizes the conditional likelihood. For this purpose, genetic algorithms are used. The addressed issues are challenging problems and are substantial for developing an audio-visual speech recognition framework that can maximize the information gather about the words uttered and minimize the impact of noise.

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49

Chen, Bidong. "Audio recognition with distributed wireless sensor networks". Thesis, 2010. http://hdl.handle.net/1828/2683.

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Recent technique advances have made sensor nodes to be smaller, cheaper and more powerful. Compared with traditional centralized sensing systems, wireless sensor networks are very easy to deploy and can be deployed densely. They have a better sensing coverage and provide more reliable information delivery. Those advantages make wireless sensor networks very useful in a wide variety of applications. As one of active research areas, acoustic monitoring with wireless sensor networks is still new, and very few applications can recognize human voice, discriminate human speech and music, or identify individual speakers. In this thesis work, we designed and implemented an acoustic monitoring system with a wireless sensor network to classify human voice versus music. We also introduce a new, effective sound source localization method, using Root Mean Square (RMS) detected by different nodes of a wireless sensor network to estimate the speaker's location. The experimental results show that our approaches are effective. This research could form a basis for further developing speech recognition, speaker identification, even emotion detection with wireless sensor networks.

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50

Liao, Wen-Yuan, i 廖文淵. "A Study on Audio-Visual Feature Extraction for Mandarin Digit Speech Recognition". Thesis, 2009. http://ndltd.ncl.edu.tw/handle/46704732964354703864.

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博士
大同大學
資訊工程學系(所)
97
In recent years, there have been many machine speechreading systems proposed, that combine audio and visual speech features. For all such systems, the objective of these audio-visual speech recognizers is to improve recognition accuracy, particularly in difficult condition. This thesis presents a Mandarin audio-visual recognition system that has better recognition rate in noisy condition as well as speech spoken with emotional condition. We first extract the visual features of the lips, including geometric and motion features. These features are very important to the recognition system especially in noisy condition or with emotional effects. The motion features are obtained by applying an automatic face feature extractor followed by a fast motion feature extractor. We compare the performance when the system using motion and geometric features. In this recognition system, we propose to use the weighted-discrete KNN as the classifier and compare the results with two popular classifiers, the GMM and HMM, and evaluate their performance by applying to a Mandarin audio-visual speech corpus. We find that the WD-KNN is a suitable classifier for Mandarin speech because the monosyllable property of Mandarin and computationally inexpensive. The experimental results of different classifiers at various SNR levels are presented. The results show that using the WD-KNN classifier yields better recognition accuracy than other classifiers for the used Mandarin speech corpus. Several weighting functions were also studied for the weighted KNN based classifier, such as linear distance weighting, inverse distance weighting, rank weighting and reverse Fibonacci weighting function. The overall results have proved that WD-KNN classifier with reverse Fibonacci weighting function gets the higher recognition rate in three extended versions of KNN outperform others. Finally, we perform the emotional speech recognition experiments. The results show that it will be more robust if the visual information is included. The recognition rate of the audio-visual speech recognition system will have higher recognition rate when incorporated with the visual cues.

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