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Dissertations / Theses on the topic 'Speech waveforms'
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1
Lowry, Andrew. "Efficient structures for vector quantisation of speech waveforms." Thesis, Queen's University Belfast, 1988. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.292596.
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Carandang, Alfonso B., and n/a. "Recognition of phonemes using shapes of speech waveforms in WAL." University of Canberra. Information Sciences & Engineering, 1994. http://erl.canberra.edu.au./public/adt-AUC20060626.144432.
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Generating a phonetic transcription of the speech waveform is one method
which can be applied to continuous speech recognition. Current methods of labelling a
speech wave involve the use of techniques based on spectrographic analysis. This paper
presents a computationally simple method by which some phonemes can be identified
primarily by their shapes.
Three shapes which are regularly manifested by three phonemes were examined
in utterances made by a number of speakers. Features were then devised to recognise
their patterns using finite state automata combined with a checking mechanism. These
were implemented in the Wave Analysis Language (WAL) system developed at the
University of Canberra and the results showed that the phonemes can be recognised
with high accuracy. The resulting shape features have also demonstrated a degree of
speaker independence and context dependency.
3
Deivard, Johannes. "How accuracy of estimated glottal flow waveforms affects spoofed speech detection performance." Thesis, Mälardalens högskola, Akademin för innovation, design och teknik, 2020. http://urn.kb.se/resolve?urn=urn:nbn:se:mdh:diva-48414.
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In the domain of automatic speaker verification, one of the challenges is to keep the malevolent people out of the system. One way to do this is to create algorithms that are supposed to detect spoofed speech. There are several types of spoofed speech and several ways to detect them, one of which is to look at the glottal flow waveform (GFW) of a speech signal. This waveform is often estimated using glottal inverse filtering (GIF), since, in order to create the ground truth GFW, special invasive equipment is required. To the author’s knowledge, no research has been done where the correlation of GFW accuracy and spoofed speech detection (SSD) performance is investigated. This thesis tries to find out if the aforementioned correlation exists or not. First, the performance of different GIF methods is evaluated, then simple SSD machine learning (ML) models are trained and evaluated based on their macro average precision. The ML models use different datasets composed of parametrized GFWs estimated with the GIF methods from the previous step. Results from the previous tasks are then combined in order to spot any correlations. The evaluations of the different methods showed that they created GFWs of varying accuracy. The different machine learning models also showed varying performance depending on what type of dataset that was being used. However, when combining the results, no obvious correlations between GFW accuracy and SSD performance were detected. This suggests that the overall accuracy of a GFW is not a substantial factor in the performance of machine learning-based SSD algorithms.
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Bamini, Praveen Kumar. "FPGA-based Implementation of Concatenative Speech Synthesis Algorithm." [Tampa, Fla.] : University of South Florida, 2003. http://purl.fcla.edu/fcla/etd/SFE0000187.
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Choy, Eddie L. T. "Waveform interpolation speech coder at 4 kbs." Thesis, McGill University, 1998. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=20901.
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Speech coding at bit rates near 4 kbps is expected to be widely deployed in applications such as visual telephony, mobile and personal communications. This research focuses on developing a speech coder based on the waveform interpolation (WI) scheme, with an attempt to deliver near toll-quality speech at rates around 4 kbps. A WI coder has been simulated in floating-point using the C programming language. The high performance of the WI model has been confirmed by subjective listening tests in which the unquantized coder outperforms the 32 kbps G.726 standard (ADPCM) 98% of the time under clean input speech conditions; the reconstructed speech is perceived to be essentially indistinguishable from the original. When fully quantized, the speech quality of the WI coder at 4.25 kbps has been judged to be equivalent to or better than that of G.729 (the ITU-T toll-quality 8 kbps standard) for 45% of the test sentences. Further refinements of the quantization techniques are warranted to bring the coder closer to the toll-quality benchmark. Yet, the existing implementation has produced good quality coded speech with a high degree of intelligibility and naturalness when compared to the conventional coding schemes operating in the neighbourhood of 4 kbps.
6
Leong, Michael. "Representing voiced speech using prototype waveform interpolation for low-rate speech coding." Thesis, McGill University, 1992. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=56796.
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In recent years, research in narrow-band digital speech coding has achieved good quality speech coders at low rates of 4.8 to 8.0 kb/s. This thesis examines the method proposed by W. B. Kleijn called prototype waveform interpolation (PWI) for coding the voiced sections of speech efficiently to achieve a coder below 4.8 kb/s while maintaining, even improving, the perceptual quality of current coders.In examining the PWI method, it was found that although the method generally works very well there are occasional sections of the reconstructed voiced speech where audible distortion can be heard, even when the prototypes are not quantized. The research undertaken in this thesis focuses on the fundamental principles behind modelling voiced speech using PWI instead of focusing on bit allocation for encoding the prototypes. Problems in the PWI method are found that may be have been overlooked as encoding error if full encoding were implemented.
Kleijn uses PWI to represent voiced sections of the excitation signal which is the residual obtained after the removal of short-term redundancies by a linear predictive filter. The problem with this method is that when the PWI reconstructed excitation is passed through the inverse filter to synthesize the speech undesired effects occur due to the time-varying nature of the filter. The reconstructed speech may have undesired envelope variations which result in audible warble.
This thesis proposes an energy fixup to smoothen the synthesized speech envelope when the interpolation procedure fails to provide the smooth linear result that is desired. Further investigation, however, leads to the final proposal in this thesis that PWI should he performed on the clean speech signal instead of the excitation to achieve consistently reliable results for all voiced frames.
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Choy, Eddie L. T. "Waveform interpolation speech coder at 4 kb/s." Thesis, National Library of Canada = Bibliothèque nationale du Canada, 1998. http://www.collectionscanada.ca/obj/s4/f2/dsk1/tape11/PQDD_0028/MQ50596.pdf.
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Davis, Andrew J. "Waveform coding of speech and voiceband data signals." Thesis, University of Liverpool, 1988. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.232946.
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Eide, Ellen Marie. "A linguistic feature representation of the speech waveform." Thesis, Massachusetts Institute of Technology, 1993. http://hdl.handle.net/1721.1/12510.
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Thesis (Ph. D.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 1993.Includes bibliographical references (leaves 95-97).
by Ellen Marie Eide.
Ph.D.
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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 boutWhile 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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ARAUJO, ANTONIO MARCOS DE LIMA. "ANALYSIS OF WAVEFORM CODERS FOR SPEECH AND DATA SIGNALS." PONTIFÍCIA UNIVERSIDADE CATÓLICA DO RIO DE JANEIRO, 1986. http://www.maxwell.vrac.puc-rio.br/Busca_etds.php?strSecao=resultado&nrSeq=9246@1.
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O trabalho examina o comportamento de Codificadores de
forma de onda operando a 32,56 e 64kbit/s para transmissão
digital de sinais de voz e de sinais de dados PSK-8 a 4800
bit/s e QAM-16 a 9600 bit/s. A partir de uma análise
detalhada dos diversos sistemas, tanto em canal ideal como
um canal ruidoso, é verificada a necessidade de se fazer
uma identificação do tipo de sinal. De modo a permitir sua
codificação de forma mais eficiente. É, então, proposta e
avaliada a utilização de uma técnica de identificação
estatística de sinais de voz e dados, em codificadores de
forma de onda. A incorporação desta técnica ao sistema
ADPCM a 32 kbit/s recomendado pelo CCITT permite uma
melhoria do desempenho para sinais de dados, sem com isso
alterar sua eficiência para sinais de voz.This thesis evaluates the performance of waveform coders at 32,56 and 64kbit/s for digital transmission of speech signal and 4800 bit/s PSK-8 and 9600 bit/s QAM-16 voiceband data signas. A detailed analysis of the systems is carried out both under ideal and noisy channel conditions. From this analysis it was found that a scheme which accurately distinguishes the two classes of signals, would allow a more efficient encoding procedure. A method of statistical identification of speech and data signals is proposed and its use in wakeform coders is, then, analysed. The incorporation of this method into the 32 kbit/s ADPCM system recommended by CCITT provides an improvement in performance for data signals, without sacrificing its efficiency for speech signal.
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Yaghmaie, Khashayar. "Prototype waveform interpolation based low bit rate speech coding." Thesis, University of Surrey, 1997. http://epubs.surrey.ac.uk/843152/.
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Advances in digital technology in the last decade have motivated the development of very efficient and high quality speech compression algorithms. While in the early low bit rate coding systems, the main target was production of intelligible speech at low information rates, expansion of new applications such as mobile satellite systems increased the demand for high quality speech at lowest possible bit rates. This resulted in the development of efficient parametric models for speech production system. These models were the basis of powerful speech compression algorithms such as CELP and Multiband excitation. CELP is a very efficient algorithm at medium bit rates and has achieved almost toll quality at 8 kb/s. However, the performance of CELP rapidly reduces at bit rates below 4.8 kb/s. The sinusoidal based coding algorithms and in particular multiband excitation technique have proved their abilities in producing high quality speech at bit rates below 5 kb/s. In recent years, another efficient speech compression algorithm called prototype waveform interpolation (PWI) has emerged. PWI presented a novel model which proved to be very efficient in removing redundant information from speech. While the early PWI systems produced high quality speech at bit rates around 3.5 kb/s, its latest versions produce an even higher quality at the bit rates as low as 2.4 kb/s. The key to the success of PWI is the approach it exploits in reducing the distortion associated with low bit rate coding algorithms. However, the price for this achievement is a very high computational demand which has been the main hurdle in its real time applications. The aim of the research in this thesis is the development of low complexity PWI systems without sacrificing the high quality. While the target of the majority of PWI systems is efficient coding of the excitation signal in the LP model of speech, this research focuses on exploiting PWI to directly encode the original speech. In the first part of the thesis, basic techniques in low bit rate speech coding are described and proper tools are developed to be exploited in a PWI based coding system. In the second part, the original PWI algorithm operating in the LP residual domain is briefly explained and application of PWI in speech domain is introduced as a method to cope with problems associated with the original PWI. To demonstrate the abilities of this approach, various coding schemes operating in the range of 1.85 to 2.95 kb/s are developed. In the final stage, a new technique which combines the two powerful low bit rate coding techniques, i.e multiband excitation and PWI, is developed to produce high quality synthetic speech at 2.6 kb/s.
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Khan, Mohammad M. A. "Coding of excitation signals in a waveform interpolation speech coder." Thesis, McGill University, 2001. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=32961.
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The goal of this thesis is to improve the quality of the Waveform Interpolation (WI) coded speech at 4.25 kbps. The quality improvement is focused on the efficient coding scheme of voiced speech segments, while keeping the basic coding format intact. In the WI paradigm voiced speech is modelled as a concatenation of the Slowly Evolving pitch-cycle Waveforms (SEW). Vector quantization is the optimal approach to encode the SEW magnitude at low bit rates, but its complexity imposes a formidable barrier.Product code vector quantizers (PC-VQ) are a family of structured VQs that circumvent the complexity obstacle. The performance of product code VQs can be traded off against their storage and encoding complexity. This thesis introduces split/shape-gain VQ---a hybrid product code VQ, as an approach to quantize the SEW magnitude. The amplitude spectrum of the SEW is split into three non-overlapping subbands. The gains of the three subbands form the gain vector which are quantized using the conventional Generalized Lloyd Algorithm (GLA). Each shape vector obtained by normalizing each subband by its corresponding coded gain is quantized using a dimension conversion VQ along with a perceptually based bit allocation strategy and a perceptually weighted distortion measure. At the receiver, the discontinuity of the gain contour at the boundary of subbands introduces buzziness in the reconstructed speech. This problem is tackled by smoothing the gain versus frequency contour using a piecewise monotonic cubic interpolant. Simulation results indicate that the new method improves speech quality significantly.
The necessity of SEW phase information in the WI coder is also investigated in this thesis. Informal subjective test results demonstrate that transmission of SEW magnitude encoded by split/shape-gain VQ and inclusion of a fixed phase spectrum drawn from a voiced segment of a high-pitched male speaker obviates the need to send phase information.
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Choi, Hung Bun. "Pitch synchronous waveform interpolation for very low bit rate speech coding." Thesis, University of Liverpool, 1997. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.243264.
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Pollard, Matthew Peter. "Waveform interpolation methods for pitch and time-scale modification of speech." Thesis, University of Liverpool, 1997. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.263905.
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Smith, Lloyd A. (Lloyd Allen). "Speech Recognition Using a Synthesized Codebook." Thesis, University of North Texas, 1988. https://digital.library.unt.edu/ark:/67531/metadc332203/.
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Speech sounds generated by a simple waveform synthesizer were used to create a vector quantization codebook for use in speech recognition. Recognition was tested over the TI-20 isolated word data base using a conventional DTW matching algorithm. Input speech was band limited to 300 - 3300 Hz, then passed through the Scott Instruments Corp. Coretechs process, implemented on a VET3 speech terminal, to create the speech representation for matching. Synthesized sounds were processed in software by a VET3 signal processing emulation program. Emulation and recognition were performed on a DEC VAX 11/750.
The experiments were organized in 2 series. A preliminary experiment, using no vector quantization, provided a baseline for comparison.
The original codebook contained 109 vectors, all derived from 2 formant synthesized sounds. This codebook was decimated through the course of the first series of experiments, based on the number of times each vector was used in quantizing the training data for the previous experiment, in order to determine the smallest subset of vectors suitable for coding the speech data base. The second series of experiments altered several test conditions in order to evaluate the applicability of the minimal synthesized codebook to conventional codebook training.
The baseline recognition rate was 97%. The recognition rate for synthesized codebooks was approximately 92% for sizes ranging from 109 to 16 vectors. Accuracy for smaller codebooks was slightly less than 90%. Error analysis showed that the primary loss in dropping below 16 vectors was in coding of voiced sounds with high frequency second formants. The 16 vector synthesized codebook was chosen as the seed for the second series of experiments.
After one training iteration, and using a normalized distortion score, trained codebooks performed with an accuracy of 95.1%. When codebooks were trained and tested on different sets of speakers, accuracy was 94.9%, indicating that very little speaker dependence was introduced by the training.
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Ghaidan, Khaldoon A. "A study of the application of modern techniques to speech waveform analysis." Thesis, Loughborough University, 1986. https://dspace.lboro.ac.uk/2134/28015.
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Spectrograms are perhaps the most commonly used method for studying the characteristics of speech waveforms. Producing a spectrogram can conveniently be divided into two parts, the analysis and the display, and this thesis describes a study of both these aspects.
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Bliūdžius, Mindaugas. "Skaitmeninių kalbos įrašų glaudinimo metodai." Master's thesis, Lithuanian Academic Libraries Network (LABT), 2004. http://vddb.library.lt/obj/LT-eLABa-0001:E.02~2004~D_20040529_122424-17577.
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The past three decades has witnessed substantial progress towards the application of low-rate speech coders to civilian and military communications as well as computer-related voice applications. Central to this progress has been the development of new speech coders capable of producing high-quality speech at low data rates. Most of these coders incorporate mechanisms to: represent the spectral properties of speech, provide for speech waveform matching, and "optimize" the coder's performance for the human ear. A number of these coders have already been adopted in national and international cellular telephony standards. The objective of this paper is to provide a tutorial overview of speech coding methodologies with emphasis on those algorithms that are part of the recent low-rate standards for voice applications. Although the emphasis is on the new low-rate coders, we attempt to provide a comprehensive survey by covering some of the traditional methodologies as well. The paper starts with a historical perspective and continues with a brief discussion on the speech properties and performance measures. Then I proceed with descriptions of waveform coders, linear predictive vocoders, and analysis-by-synthesis linear predictive coders. At the end the system for computer-based stenographing is presented. Quality research and ways how to improve this system will be provided.
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Beall, Jeffery C. "Stored waveform adaptive motor control." Thesis, Virginia Tech, 1986. http://hdl.handle.net/10919/45746.
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This study investigates an adaptive control scheme designed to maintain accurate motor speed control in spite of high-frequency periodic variations in load torque, load inertia, and motor parameters. The
controller adapts, stores and replays a schedule of torques to be delivered at discrete points throughout the periodic load cycle. The controller also adapts to non-periodic changes in load conditions which occur over several load cycles and contains inherent integrator control action to drive speed error to zero. Using computer simulations, the control method was successfully applied to a 3Φ synchronous motor and a permanent magnet D.C. motor. The D.C. motor (or A.C. servo-motor) controller has superior characteristics and this system performance was compared to P, PI and PID control for two severe load cases - a periodic step load and a four-bar linkage load. Simulation studies showed the schedule control method to be stable and in comparison to the PID controller to have 1) nearly the same speed of response but without the overshoot found in PID control, 2) nearly the same mean speed error (~ O), 3) 12-50 times better reduction in speed fluctuation, and 4) the schedule controller gains were much easier to find than PID gains for this low-order, highly responsive system.Master of Science
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Pelteku, Altin E. "Development of an electromagnetic glottal waveform sensor for applications in high acoustic noise environments." Link to electronic thesis, 2004. http://www.wpi.edu/Pubs/ETD/Available/etd-0114104-142855/.
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Thesis (M.S.)--Worcester Polytechnic Institute.Keywords: basis functions; perfectly matched layers; PML; neck model; parallel plate resonator; finite element; circulator; glottal waveform; multi-transmission line; dielectric properties of human tissues; radiation currents; weighted residuals; non-acoustic sensor. Includes bibliographical references (p. 104-107).
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Keenaghan, Kevin Michael. "A Novel Non-Acoustic Voiced Speech Sensor Experimental Results and Characterization." Link to electronic thesis, 2004. http://www.wpi.edu/Pubs/ETD/Available/etd-0114104-144946/.
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Nehl, Albert Henry. "Investigation of techniques for high speed CMOS arbitrary waveform generation." PDXScholar, 1990. https://pdxscholar.library.pdx.edu/open_access_etds/4109.
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Today a growing number of applications in design engineering, production and environmental testing, and system service require specific analog waveforms and digital patterns. Such requirements are neither satisfactorily nor easily met by the use of standard function or single purpose, custom generators.
Traditional methods of waveform generation suffer from undesirable complexity or mediocre performance and are otherwise limited. For the majority of arbitrary waveform generation applications, including medical engineering, modal analysis and electronic engineering, direct digital synthesis techniques are satisfactory. Direct digital synthesis, based generally on periodic retrieval of predetermined amplitude values, may be used to 2 generate such waveforms. Within the limits imposed by the system's maximum sample rate and the Nyquist criteria, any waveform may be produced using these techniques.
The objective of this inquiry, within a particular set of constraints, is to extend the cost/performance envelope of direct digital synthesis techniques for the generation of arbitrary waveforms. Performance is enhanced, particularly in the areas of output bandwidth and signal purity.
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Joseph, Andrew Paul. "Assessing the effects of GMAW-Pulse parameters on arc power and weld heat input /." Connect to resource, 2001. http://rave.ohiolink.edu/etdc/view?acc%5Fnum=osu1130521651.
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Elangovan, Saravanan, Jerry L. Cranford, Letitia Walker, and Andrew Stuart. "A Comparison of the Mismatch Negativity and a Differential Waveform Response." Digital Commons @ East Tennessee State University, 2005. https://dc.etsu.edu/etsu-works/1556.
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A mismatch negativity response (MMN) and a new differential waveform were derived in an effort to evaluate a neural refractory or recovery effect in adult listeners. The MMN was elicited using oddball test runs in which the standard and deviant stimuli differed in frequency. To derive the differential waveform, the same standard and deviant stimuli were presented alone. MMN responses were obtained by subtracting the averaged responses to standards from the deviants. The differential waveforms were obtained by subtracting the averaged responses to standards presented alone from deviants presented alone. Scalp topography for the MMN and differential waveforms were similar. A significant (p
Se obtuvo una respuesta de negatividad desigual (MMN) y una nueva onda ?diferencial? en un esfuerzo por evaluar un efecto neural refractario o de recuperación en sujetos adultos. La MMN fue generada utilizando cursos peculiares de prueba en los que el estimulo estándar y el alterado tenían frecuencias diferentes. Para derivar la onda diferencial, se presentaron el mismo estímulo estándar y el alterado en forma aislada. Las respuestas MMN se obtuvieron restando las respuestas promediadas estándar de las alteradas. Las formas de onda diferenciales se obtuvieron restando las respuestas promediadas a estímulos estándar presentados aisladamente, de los estímulos alterados presentados también en forma aislada. La topografía craneana de los MMN y las onda diferenciales fueron similares. Se encontraron correlaciones positivas y negativas significativas (p
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Torres, Juan Félix. "Estimation of glottal source features from the spectral envelope of the acoustic speech signal." Diss., Georgia Institute of Technology, 2010. http://hdl.handle.net/1853/34736.
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Speech communication encompasses diverse types of information, including phonetics, affective state, voice quality, and speaker identity. From a speech production standpoint, the acoustic speech signal can be mainly divided into glottal source and vocal tract components, which play distinct roles in rendering the various types of information it contains. Most deployed speech analysis systems, however, do not explicitly represent these two components as distinct entities, as their joint estimation from the acoustic speech signal becomes an ill-defined blind deconvolution problem. Nevertheless, because of the desire to understand glottal behavior and how it relates to perceived voice quality, there has been continued interest in explicitly estimating the glottal component of the speech signal. To this end, several inverse filtering (IF) algorithms have been proposed, but they are unreliable in practice because of the blind formulation of the separation problem. In an effort to develop a method that can bypass the challenging IF process, this thesis proposes a new glottal source information extraction method that relies on supervised machine learning to transform smoothed spectral representations of speech, which are already used in some of the most widely deployed and successful speech analysis applications, into a set of glottal source features. A transformation method based on Gaussian mixture regression (GMR) is presented and compared to current IF methods in terms of feature similarity, reliability, and speaker discrimination capability on a large speech corpus, and potential representations of the spectral envelope of speech are investigated for their ability represent glottal source variation in a predictable manner. The proposed system was found to produce glottal source features that reasonably matched their IF counterparts in many cases, while being less susceptible to spurious errors. The development of the proposed method entailed a study into the aspects of glottal source information that are already contained within the spectral features commonly used in speech analysis, yielding an objective assessment regarding the expected advantages of explicitly using glottal information extracted from the speech signal via currently available IF methods, versus the alternative of relying on the glottal source information that is implicitly contained in spectral envelope representations.
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Liu, Lup Shun Nelson Carleton University Dissertation Engineering Electronics. "Sensitivity analysis and optimization of high-speed VLSI interconnects using asymptotic waveform evaluation." Ottawa, 1993.
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Bhatta, Debesh. "Algorithms and methodology for incoherent undersampling based acquisition of high speed signal waveforms using low cost test instrumentation." Diss., Georgia Institute of Technology, 2014. http://hdl.handle.net/1853/54262.
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The objective of this research is to develop and demonstrate low-complexity, robust, frequency-scalable, wide-band waveform acquisition techniques for testing high speed com-
munication systems.
High resolution waveform capture is a versatile testing tool that enables flexible test strategies. However, waveform capture at high data rates requires costly hardware because the increased bandwidth of the signal waveform leads to an increase in the sampling rate requirement, cost of front-end components, and sensitivity to phase errors in traditional (source) synchronous Nyquist-rate tester architectures. The hardware cost and complexity of wide-band waveform acquisition systems can, however, be significantly reduced by using (trigger-free) incoherent undersampling to achieve reduced sampling rates and robustness to phase errors in signal paths. Reducing the hardware cost of such a system using incoherent undersampling requires increased signal processing at the back end.
This research proposes computationally-efficient, time-domain waveform reconstruction algorithms to improve both performance, and scope of existing incoherent undersampling-
based test instrumentation. Supporting hardware architectures are developed to extend the application of incoherent undersampling-based waveform acquisition techniques to linearity testing of high-speed radio-frequency components without any synchronization between the signals involved, and to the acquisition of wide-band signals beyond the track-and-hold
bandwidth barrier of the traditional incoherent undersampling architectures, using multi-channel bandwidth interleaving. The bandwidth is extended in a source-incoherent framework by using mixers to down convert high-frequency signal components to base band followed by digitization using undersampling, and back-end signal processing to reconstruct the original wide-band signal from multiple band-pass components.
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Tsutsumi, Monike. "Avaliação da videolaringoscopia de alta velocidade de sujeitos normais." Universidade de São Paulo, 2015. http://www.teses.usp.br/teses/disponiveis/82/82131/tde-28032016-142207/.
Full textAbstract:
Estudos utilizando imagens laríngeas de sujeitos normais captadas por videolaringoscopia de alta velocidade revelam o uso de diversas ferramentas e métricas em diferentes populações. No entanto, é evidente a escassez de normatizações operacionais e de parâmetros vocais de referência. Os objetivos desse estudo foram obter parâmetros da dinâmica vocal utilizando ferramentas computacionais de uso corrente pelo Grupo de Pesquisa em Engenharia Médica (GPEM - CNPq) e caracterizar o padrão vibratório das pregas vocais de sujeitos normais por meio das ondas da área glótica e quimografia de alta velocidade do utilizando Videolaringoscopia de alta velocidade. Metodologia: A partir de imagens laríngeas captadas pela videolaringoscopia de alta velocidade foram extraídos os parâmetros quantitativos: i) tempos de fases e período total do ciclo vibratório das pregas vocais das ondas da área glótica, ii) tempos de fases e período total do ciclo vibratório das pregas vocais da quimografia de alta velocidade, iii) coeficientes (de abertura, de fechamento e de velocidade). Além disso, foram analisados os parâmetros qualitativos das ondas da área glótica e da quimografia de alta velocidade de acordo com a aplicação de protocolos elaborados para classificação de padrões visuais. Resultados: Das ondas da área glótica foram obtidos os valores médios, em milissegundos, de fase fechada: feminino=0.83 e masculino= 2.47; de abertura: feminino= 2.43 e masculino= 2.95; de fechamento: feminino=2.08 e masculino= 2.53; aberta: feminino= 6.15 e masculino= 6.18, período total do ciclo vibratório: feminino=6.98 e masculino= 8.65; coeficientes: de fechamento: feminino=0.14 e masculino=0.29, de abertura: feminino=0.85 e masculino=0.70, de velocidade: feminino=1.16 e masculino=1.19, além de 73% dos traçados apresentarem sinal periódico. Quanto à quimografia de alta velocidade os parâmetros quantitativos obtidos foram: fase fechada: feminino=1.75 e masculino=3.32, de abertura: feminino= 1.47 e masculino= 2.32; de fechamento: feminino= 1.51 e masculino= 2.22; aberta: feminino= 2.91 e masculino= 4.56, e período total do ciclo vibratório: feminino= 4.67 e masculino= 7.89. Os coeficientes obtidos foram: de fechamento: feminino= 0.37 e masculino= 0.42.; de abertura: feminino= 0.62 e masculino= 0.57; e de velocidade: feminino= 1.02 e masculino= 1.12. 59% de simetria em amplitude e 54% de assimetria de fase foram encontrados no traçado da quimografia de ata velocidade de sujeitos normais. Conclusão: A partir do uso de ferramentas computacionais específicas para analisar imagens laríngeas da videolaringoscopia de alta velocidade foi possível obter parâmetros quantitativos e qualitativos das ondas da área glótica e quimografia de alta velocidade de sujeitos normais e obter dados de referência quanto à normalidade para futuros estudos.Several studies using laryngeal images from high-speed videolaryngoscopy of normal subjects reveals the diversity of tools and metrics used for different population. However, shortage of operational standardization and references of vocal fold parameters are evident. The main objectives of this study were to obtain parameters of vocal dynamics using computational tools of Medical Engineering Research Group (GPEM - CNPq) and to characterize the vocal fold\'s vibration pattern of normal subjects using glottal area waveforms and high-speed kymography. Methods: From laryngeal images of high-speed videolaryngoscopy we extracted the following quantitative parameters: i) phase time of glottal area waveforms, ii) phase time of vibratory cicle\'s total period, iii) quocients of high-speed kymography. Furthermore, qualitative parameters of glottal area waveform were analyzed according to visual pattern protocol. Results: Media values of glottal area waveforms, in milliseconds, of closed phase: female=0.83 and male= 2.47; opening phase: female= 2.43 and male= 2.95; closing phase: female= 2.08 and male= 2.53; opened phase: female=6.15 and male= 6.18, vibratory cicle of total period: female= 6.98 and male= 8.65, closing quotient: female= 0.14 and male= 0.29; opening quotient: female= 0.85 and male= 0.70; speed quotient: female= 1.16 and male= 1.19, besides 73% showed periodic signal. As the high- speed kymography the quantitative parameters obtained were: closed phase: female= 1.75 and male= 3.32; opening phase: female= 1.47 and male= 2.32; closing phase: female= 1.51 and male= 2.22; opened phase: female= 2.91 and male= 4.56, and vibratory cicie of total period: female= 4.67 and male= 7.89. The quotients obtained were: closing quotient: female= 0.37and male= 0.42; opening quotient: female= 0.62 and male= 0.57; speed quotient: female= 1.02 and male= 1.12. 59% amplitude symmetry and 54% phase asymmetry were obtained in the high- speed kymography of normal subjects. Conclusion: using specific computational tools to analyse high-speed laryngeal images we obtained quantitative and qualitative parameters of glottal area waveforms and high-speed kymography that can be used as a standard reference data for normal subjects.
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Schulz, Yvonne Katrin [Verfasser], Stefan [Akademischer Betreuer] Kniesburges, and Stefan [Gutachter] Kniesburges. "Parameter analysis of the Glottal Area Waveform based on high-speed recordings within a synthetic larynx model / Yvonne Katrin Schulz ; Gutachter: Stefan Kniesburges ; Betreuer: Stefan Kniesburges." Erlangen : Friedrich-Alexander-Universität Erlangen-Nürnberg (FAU), 2020. http://d-nb.info/120337769X/34.
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Baker, Katherine Louise. "Cognitive Evoked Auditory Potentials and Neuropsychological Measures Following Concussion in College Athletes." Miami University / OhioLINK, 2008. http://rave.ohiolink.edu/etdc/view?acc_num=miami1209744334.
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Narayanan, G. "Synchronised Pulsewidth Modulation Strategies Based On Space Vector Approach For Induction Motor Drives." Thesis, Indian Institute of Science, 1999. http://hdl.handle.net/2005/139.
Full textAbstract:
In high power induction motor drives, the switching frequency of the inverter is quite low due to the high losses in the power devices. Real-time PWM strategies, which result in reduced harmonic distortion under low switching frequencies and have maximum possible DC bus utilisation, are developed for such drives in the present work.
The space vector approach is taken up for the generation of synchronised PWM waveforms with 3-Phase Symmetry, Half Wave Symmetry and Quarter Wave Symmetry, required for high-power drives. Rules for synchronisation and the waveform symmetries are brought out. These rules are applied to the conventional and modified forms of space vector modulation, leading to the synchronised conventional space vector strategy and the Basic Bus Clamping Strategy-I, respectively. Further, four new synchronised, bus-clamping PWM strategies, namely Asymmetric Zero-Changing Strategy, Boundary Sampling Strategy-I, Basic Bus Clamping Strategy-II and Boundary Sampling Strategy-II, are proposed. These strategies exploit the flexibilities offered by the space vector approach like double-switching of a phase within a subcycle, clamping of two phases within a subcycle etc. It is shown that the PWM waveforms generated by these strategies cannot be generated by comparing suitable 3-phase modulating waves with a triangular carrier wave.
A modified two-zone approach to overmodulation is proposed. This is applied to the six synchronised PWM strategies, dealt with in the present work, to extend the operation of these strategies upto the six-step mode. Linearity is ensured between the magnitude of the reference and the fundamental voltage generated in the whole range of modulation upto the six-step mode. This is verified experimentally.
A suitable combination of these strategies leads to a significant reduction in the harmonic distortion of the drive at medium and high speed ranges over the conventional space vector strategy. This reduction in harmonic distortion is demonstrated, theoretically as well as experimentally, on a constant V/F drive of base frequency 50Hz for three values of maximum switching frequency of the inverter, namely 450Hz, 350Hz and 250Hz.
Based on the notion of stator flux ripple, analytical closed-form expressions are derived for the harmonic distortion due to the different PWM strategies. The values of harmonic distortion, computed based on these analytical expressions, compare well with those calculated based on Fourier analysis and those measured experimentally.
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LIN, YAN-JUN, and 林妍君. "Speech waveform coding based on vector quantization." Thesis, 1986. http://ndltd.ncl.edu.tw/handle/88277127723707099722.
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Biskup, John Fredrick. "Applications of the speedy delivery waveform." Thesis, 2003. http://hdl.handle.net/2152/29809.
Full textAbstract:
The Speedy Delivery (SD) waveform was introduced in patent US 6,441,695 B1 issued August 27, 2002 to the inventor Dr. Robert Flake. In the most basic form, the SD boundary condition is an exponential, D⋅e [superscript α⋅t] . The propagating waveform is described by an analytic, closed form solution of the wave equation in lossy media and has several very special properties. The most surprising property is that the leading edge of the waveform propagates with attenuation but without distortion. The lack of distortion occurs even in lossy transmission media with frequency dependent parameters. This is unlike any other known pulse shape. Additionally, varying the waveforms parameter, α, can vary the propagation velocity and the attenuation of the waveform. Because the exponential waveform is unbounded it cannot continue indefinitely and must be truncated and closed by a non-SD closing edge. This dissertation discusses the transmission behavior and two applications of truncated SD waveforms. A brief analysis of SD propagation in lossy transmission lines is presented and some practical considerations associated with truncating the SD waveforms are addressed. The parameters needed to describe the propagation of the SD waveform are defined and techniques for determining their values are presented. Finally, examples applying these truncated SD waveforms to time domain reflectometry and Communication Technology are presented.text
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"Unit selection and waveform concatenation strategies in Cantonese text-to-speech." 2005. http://library.cuhk.edu.hk/record=b5892349.
Full textAbstract:
Oey Sai Lok.Thesis (M.Phil.)--Chinese University of Hong Kong, 2005.
Includes bibliographical references.
Abstracts in English and Chinese.
Chapter 1. --- Introduction --- p.1
Chapter 1.1 --- An overview of Text-to-Speech technology --- p.2
Chapter 1.1.1 --- Text processing --- p.2
Chapter 1.1.2 --- Acoustic synthesis --- p.3
Chapter 1.1.3 --- Prosody modification --- p.4
Chapter 1.2 --- Trends in Text-to-Speech technologies --- p.5
Chapter 1.3 --- Objectives of this thesis --- p.7
Chapter 1.4 --- Outline of the thesis --- p.9
References --- p.11
Chapter 2. --- Cantonese Speech --- p.13
Chapter 2.1 --- The Cantonese dialect --- p.13
Chapter 2.2 --- Phonology of Cantonese --- p.14
Chapter 2.2.1 --- Initials --- p.15
Chapter 2.2.2 --- Finals --- p.16
Chapter 2.2.3 --- Tones --- p.18
Chapter 2.3 --- Acoustic-phonetic properties of Cantonese syllables --- p.19
References --- p.24
Chapter 3. --- Cantonese Text-to-Speech --- p.25
Chapter 3.1 --- General overview --- p.25
Chapter 3.1.1 --- Text processing --- p.25
Chapter 3.1.2 --- Corpus based acoustic synthesis --- p.26
Chapter 3.1.3 --- Prosodic control --- p.27
Chapter 3.2 --- Syllable based Cantonese Text-to-Speech system --- p.28
Chapter 3.3 --- Sub-syllable based Cantonese Text-to-Speech system --- p.29
Chapter 3.3.1 --- Definition of sub-syllable units --- p.29
Chapter 3.3.2 --- Acoustic inventory --- p.31
Chapter 3.3.3 --- Determination of the concatenation points --- p.33
Chapter 3.4 --- Problems --- p.34
References --- p.36
Chapter 4. --- Waveform Concatenation for Sub-syllable Units --- p.37
Chapter 4.1 --- Previous work in concatenation methods --- p.37
Chapter 4.1.1 --- Determination of concatenation point --- p.38
Chapter 4.1.2 --- Waveform concatenation --- p.38
Chapter 4.2 --- Problems and difficulties in concatenating sub-syllable units --- p.39
Chapter 4.2.1 --- Mismatch of acoustic properties --- p.40
Chapter 4.2.2 --- "Allophone problem of Initials /z/, Id and /s/" --- p.42
Chapter 4.3 --- General procedures in concatenation strategies --- p.44
Chapter 4.3.1 --- Concatenation of unvoiced segments --- p.45
Chapter 4.3.2 --- Concatenation of voiced segments --- p.45
Chapter 4.3.3 --- Measurement of spectral distance --- p.48
Chapter 4.4 --- Detailed procedures in concatenation points determination --- p.50
Chapter 4.4.1 --- Unvoiced segments --- p.50
Chapter 4.4.2 --- Voiced segments --- p.53
Chapter 4.5 --- Selected examples in concatenation strategies --- p.58
Chapter 4.5.1 --- Concatenation at Initial segments --- p.58
Chapter 4.5.1.1 --- Plosives --- p.58
Chapter 4.5.1.2 --- Fricatives --- p.59
Chapter 4.5.2 --- Concatenation at Final segments --- p.60
Chapter 4.5.2.1 --- V group (long vowel) --- p.60
Chapter 4.5.2.2 --- D group (diphthong) --- p.61
References --- p.63
Chapter 5. --- Unit Selection for Sub-syllable Units --- p.65
Chapter 5.1 --- Basic requirements in unit selection process --- p.65
Chapter 5.1.1 --- Availability of multiple copies of sub-syllable units --- p.65
Chapter 5.1.1.1 --- "Levels of ""identical""" --- p.66
Chapter 5.1.1.2 --- Statistics on the availability --- p.67
Chapter 5.1.2 --- Variations in acoustic parameters --- p.70
Chapter 5.1.2.1 --- Pitch level --- p.71
Chapter 5.1.2.2 --- Duration --- p.74
Chapter 5.1.2.3 --- Intensity level --- p.75
Chapter 5.2 --- Selection process: availability check on sub-syllable units --- p.77
Chapter 5.2.1 --- Multiple copies found --- p.79
Chapter 5.2.2 --- Unique copy found --- p.79
Chapter 5.2.3 --- No matched copy found --- p.80
Chapter 5.2.4 --- Illustrative examples --- p.80
Chapter 5.3 --- Selection process: acoustic analysis on candidate units --- p.81
References --- p.88
Chapter 6. --- Performance Evaluation --- p.89
Chapter 6.1 --- General information --- p.90
Chapter 6.1.1 --- Objective test --- p.90
Chapter 6.1.2 --- Subjective test --- p.90
Chapter 6.1.3 --- Test materials --- p.91
Chapter 6.2 --- Details of the objective test --- p.92
Chapter 6.2.1 --- Testing method --- p.92
Chapter 6.2.2 --- Results --- p.93
Chapter 6.2.3 --- Analysis --- p.96
Chapter 6.3 --- Details of the subjective test --- p.98
Chapter 6.3.1 --- Testing method --- p.98
Chapter 6.3.2 --- Results --- p.99
Chapter 6.3.3 --- Analysis --- p.101
Chapter 6.4 --- Summary --- p.107
References --- p.108
Chapter 7. --- Conclusions and Future Works --- p.109
Chapter 7.1 --- Conclusions --- p.109
Chapter 7.2 --- Suggested future works --- p.111
References --- p.113
Appendix 1 Mean pitch level of Initials and Finals stored in the inventory --- p.114
Appendix 2 Mean durations of Initials and Finals stored in the inventory --- p.121
Appendix 3 Mean intensity level of Initials and Finals stored in the inventory --- p.124
Appendix 4 Test word used in performance evaluation --- p.127
Appendix 5 Test paragraph used in performance evaluation --- p.128
Appendix 6 Pitch profile used in the Text-to-Speech system --- p.131
Appendix 7 Duration model used in Text-to-Speech system --- p.132
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Xu, Wen-Long, and 許文龍. "A Mandarin Speech Synthesizer Using Time Proportionated Interpolation of Pitch Waveform." Thesis, 1996. http://ndltd.ncl.edu.tw/handle/70269225450402799343.
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Hsu, Wen-Lung, and 許文龍. "A Mandarin Speech Synthesizer Using Time Proportionated Interpolation of Pitch Waveform." Thesis, 1996. http://ndltd.ncl.edu.tw/handle/64478207895251783493.
Full textAbstract:
碩士國立臺灣科技大學
電機工程研究所
84
In this thesis, a text-to-speech system is designed and implemented on MS-Windows operating system. The 408 first-tone Mandarin syllables are adopted as the synthesis units. For the synthesis of syllable-signal, a time-domain processing method called "Time Proportionated Interpolation of Pitch Waveform (TPIPW)" is proposed. About the prosodic processing unit, a rule-based method proposed by other researchers is adopted and slightly modified here. In our method, the two parts of a syllable, i.e. the unvoiced part (e.g. voiceless consonants) and voiced parts (e.g. voiced consonants and vowels), are processed separately. The name of our method is just selected to reflect the voiced-part''s processing. By using this method, a syllable''s tone(or pitch-contour), duration, and formant- frequency height can be almost independently controlled. Especially, the duration of a syllable can be more freely changed to a value between one half and double of the original length without notable side-effects on the other two control factors. Besides, the function of increasing or decreasing formant-frequency values is provided to simulate the adjusting of vocal-track length such that the original recorded male voice can be more naturally converted to a female''s voice. For the unvoiced part, signal waveforms are classified into two classes and a method is proposed to process each class differently. This method not only synthesizes clear and intelligible signals but also support the control of duration.
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Yan, Jyh-Horng, and 顏誌宏. "The Research of Voice Waveform Reconstruction in CELP Speech Coding due to the Speech Frame''s Codewords Loss." Thesis, 1999. http://ndltd.ncl.edu.tw/handle/77877098696811301575.
Full textAbstract:
碩士國立臺北科技大學
電腦通訊與控制研究所
87
This thesis is mainly to explore the best ways to do the voice reconstruction when the transmitted speech signal in speech coding lost some frame’s codewords. In this thesis, we focus on speech coding method based on code excited linear prediction(CELP), and use the high quality low bit-rate speech vocoder -- FS1016 4.8 kbps CELP as the speech coding methods to compare the performance in the voice reconstruction under loss of frame’s codewords Considering the factors for speeding and simplicity, we use the lost frame’s codewords reconstruction as the major way for voice reconstruction under the loss of frame’s codewords. The methods for lost frame’s codewords reconstruction can be classified into two groups. One is the lost frame’s codewords reconstruction without any information of the lost frame’s codewords, the other is the lost frame’s codewords reconstruction with partial information of the lost frame’s codewords. We do experiments to investigate he best ways to do the voice reconstruction in varied situations. And we also study the importance of the parameters in a CELP frame’s codeword in voice reconstruction and improve the voice reconstruction’s quality by keeping the most important partial information in the lost frame’s codewords. Our results show that the most important parameter in the voice reconstruction due to the lost frame’s codewords is the pitch gain. By adequately keeping the parameters of pitch gain and LSF coefficients in the lost frames, we can reconstruct acceptable speech signal, evenly at high frame lost rate(30%). Finally, we implement our results as an ACM codec, and use Netmeeting to do the real-time voice transmission in the internet. We also port our programs to TI DSP chip TMS320C54x system.
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Huang, Tzu-Yun, and 黃姿云. "A Dual Complementary Acoustic Embedding Network: Mining Discriminative Characteristics from Raw-waveform for Speech Emotion Recognition." Thesis, 2019. http://ndltd.ncl.edu.tw/handle/y8zcm7.
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Chandravadan, Vora Santoshkumar. "Novel Methods For Estimation Of Static Nonlinearity Of High-Speed High-Resolution Waveform Digitizers." Thesis, 2009. http://hdl.handle.net/2005/1019.
Full textAbstract:
Analog-to-digital converter (ADC) is the main workhorse in a digital waveform recorder. Strictly speaking, an ADC is supposed to perform uniformly, irrespective of the characteristics of the signal to be acquired. However, because of certain hardware related inconsistencies, its performance declines, particularly, when acquiring non-repetitive, fast-rising, high frequency signals. The error and distortion contributed due to its declining performance, for the entire range of signals, can be comprehensively characterized by the static and dynamic nonlinearities. Actual testing of ADCs is the only way of estimating these indices. These characteristics reveal information at the microscopic level, such as bit-level aberrations, code transitions, response and settling trends, etc. These tests attain greater significance, when the digitizer is part of a reference measuring or a calibration system, because, the levels of accuracies to be achieved in such a setup may become comparable to the error introduced by the ADC. Hence, testing ADCs is a priority.
International and national standards exist for testing digital waveform recorders and ADCs. For several years, the matter related to reducing static test time of high-resolution ADCs was highlighted through many publications. A critical examination of the literature indicates the major schools-of-thought pursued so far, are, (i) refinements to ramp/triangular signal based static testing, (ii) proposals for use of alternative methods and/or test signals for static test, (iii) innovative ways of achieving a relaxation in signal source requirements and, (iv) efforts to combine static and dynamic test into a single test with an appropriate test signal. As a consequence of the literature review, objectives of the thesis were formulated. They attempt to resolve- (i) Conceive a suitable test signal for simultaneous estimation of static and dynamic nonlinearity through a single test (ii) Explore possibility of employing a low-linearity ramp signal to estimate static nonlinearity (iii) Estimating static nonlinearity by exploiting linearity property of a sine signal
• In the first part of the thesis, a method is proposed for the concurrent estimation of static and dynamic nonlinearity characteristics of an ADC, with the application of a single test signal. The novelty arises from the fact that the test signal proposed is new, and so is the concept of extracting the static and dynamic nonlinearity from the ADC output. This was achieved by conceiving a test signal, comprising of a high frequency sinusoid (which addresses the dynamic requirement), modulated by a low frequency ramp (which addresses the static requirement).
• Static characteristics of an ADC can be determined directly from the histogram-based quasi-static approach by measuring the ADC output, when excited by an ideal ramp/triangular signal of sufficiently low frequency. This approach requires only a fraction of time compared to the conventional DC test, is straightforward, easy to implement, and, in principle is an accepted method as per the revised IEEE-1057. However, the only drawback is that ramp signal sources are not ideal. Thus, nonlinearity present in the ramp signal gets superimposed on the measured ADC characteristics, which renders them, as such, unusable. The second part of the work describes a proposal to get rid of the ramp signal nonlinearity, before it is applied to the ADC. A simple method is presented which employs a low-linearity ramp signal, but yet causes only a fraction of influence on the measured ADC static characteristics.
• The third part of the thesis describes a novel method to estimate the actual static characteristics of an ADC using a low frequency sine signal, say, less than 10 Hz, by employing the histogram-based approach. It is based on the well known fact that variation of sine signal is ‘reasonably linear,’ when the angle is small. In the proposed method, the ADC under test has to be ‘fed’ with this ‘linear’ portion of the sine wave. Due to harmonics and offset in input excitation, this ‘linear’ part of the sine signal is marginally different, compared to an ideal ramp signal of equal amplitude. However, since it is a sinusoid, this difference can be accurately determined and later compensated from the measured ADC output. Thus, the corrected ADC output will correspond to the true ADC static nonlinearity. The proposed approach successfully addresses all the three concerns while estimating static linearity, i.e. it is time-efficient, excites all the ADC code-bins reasonably uniformly and tackles the source linearity issue quite nicely.
These proposals are novel, simple, easy to implement, time-efficient and importantly static nonlinearity characteristics determined from them are in good agreement with that estimated by the original DC-based technique. Implementation of each method is discussed along with experimental results, for two 8-bit digital oscilloscopes and a 10-bit real time digitizer. Further details are presented in the thesis.
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Chang, Lung-Yin, and 張龍吟. "Research on speed and quality of anodic bonding using applied voltage with various waveforms." Thesis, 2008. http://ndltd.ncl.edu.tw/handle/s656s6.
Full textAbstract:
碩士國立臺灣師範大學
工業教育學系
96
Anodic bonding technique is important and is often used in package of MEMS components. It uses ionic bond to obtain bonding results. Surface level of silicon and glass need to be very serious, and the technique belong to non-media bonding. There are important factors of bonding ratio and quality in bonding process, such as applied voltage, temperature, and type of electrode etc. In different forms of applied voltage, it causes different bonding results. The reason is described as follow: the maximum of bonding current by using constant voltage waveform would be decayed when the bonding time is increasing, but it will be kept at a high value by using variable voltage waveform and to improve bonding ratio and quality widely. The research improves that using radiate-line electrode with square voltage waveform to bonding 4 inch wafer, bonding ratio can reach 99.2% when the average voltage is 250 V, period is 8 sec, temperature is 400 ºC, and bonding time is 200 sec. In this research, we develop a novel conical frustum electrode to co-operate variable voltage waveform for anodic bonding. It not only can keep bonding current at a high value to decrease bonding time, but also can have the same bonding quality with the results of applied constant voltage. The research improves that using novel electrode with constant voltage waveform to bonding 4 inch wafer, bonding ratio can reach 99.98% when the average voltage is 800 V, temperature is 400 ºC, and bonding time is 15 sec. Using the novel electrode with square voltage waveform to bonding 4 inch wafer, bonding ratio only can reach 72.93% when the average voltage is 250 V, period is 8 sec, temperature is 400 ºC, and bonding time is 15 sec. The efficiency of bonding system is limited when using square or constant voltage waveforms to co-operate the conical frustum electrode. Although it causes output voltage can not reach the setting value in bonding process, the research still can achieve the expecting purpose.
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Zheng, Yu-Xiang, and 鄭宇翔. "Analysis of Time-domain Waveform for Stub Effect in High-Speed Digital Circuits." Thesis, 2018. http://ndltd.ncl.edu.tw/handle/p7u4d6.
Full textAbstract:
碩士中原大學
電子工程研究所
106
This paper investigates how via stubs effect the time-domain transmission (TDT) waveform, Time-Domain Reflectometry (TDR) waveform, S21, S11, and eye diagram in high speed circuit. Due to the limitations of the laboratory process, the via stub was changed to open stub. In order to understand the effect on the waveform, various parameters of the open stub are analysed. The manner in which the time-domain reflection noise affects the TDR waveform and the step voltage on the TDT waveform is investigated by lattice diagram. And in special cases, waveform overlays are used to explain the behavior of time-domain transmission waveforms. The effects of mismatch stub length due to layer change on the TDT waveform are investigated. Observe the effect on the rise time by changing the Number of open stubs. Propose the eye mask in order to create the design chart. Formulas for TDT waveform are derived by lattice diagram and the waveform overlays.
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"low bit rate speech coder based on waveform interpolation =: 基於波形預測方法的低比特率語音編碼." 1999. http://library.cuhk.edu.hk/record=b5889941.
Full textAbstract:
by Ge Gao.Thesis (M.Phil.)--Chinese University of Hong Kong, 1999.
Includes bibliographical references (leaves 101-107).
Text in English; abstracts in English and Chinese.
by Ge Gao.
Chapter 1 --- Introduction --- p.1
Chapter 1.1 --- Attributes of speech coders --- p.1
Chapter 1.1.1 --- Bit rate --- p.2
Chapter 1.1.2 --- Speech quality --- p.3
Chapter 1.1.3 --- Complexity --- p.3
Chapter 1.1.4 --- Delay --- p.4
Chapter 1.1.5 --- Channel-error sensitivity --- p.4
Chapter 1.2 --- Development of speech coding techniques --- p.5
Chapter 1.3 --- Motivations and objectives --- p.7
Chapter 2 --- Waveform interpolation speech model --- p.9
Chapter 2.1 --- Overview of speech production model --- p.9
Chapter 2.2 --- Linear prediction(LP) --- p.11
Chapter 2.3 --- Linear-prediction based analysis-by-synthesis coding(LPAS) --- p.14
Chapter 2.4 --- Sinusoidal model --- p.15
Chapter 2.5 --- Mixed Excitation Linear Prediction(MELP) model --- p.16
Chapter 2.6 --- Waveform interpolation model --- p.16
Chapter 2.6.1 --- Principles of waveform interpolation model --- p.18
Chapter 2.6.2 --- Outline of a WI coding system --- p.25
Chapter 3 --- Pitch detection --- p.31
Chapter 3.1 --- Overview of existing pitch detection methods --- p.31
Chapter 3.2 --- Robust Algorithm for Pitch Tracking(RAPT) --- p.33
Chapter 3.3 --- Modifications of RAPT --- p.37
Chapter 4 --- Development of a 1.7kbps speech coder --- p.44
Chapter 4.1 --- Architecture of the coder --- p.44
Chapter 4.2 --- Encoding of unvoiced speech --- p.46
Chapter 4.3 --- Encoding of voiced speech --- p.46
Chapter 4.3.1 --- Generation of PCW --- p.48
Chapter 4.3.2 --- Variable Dimensional Vector Quantization(VDVQ) --- p.53
Chapter 4.3.3 --- Sparse frequency representation(SFR) of speech --- p.56
Chapter 4.3.4 --- Sample selective linear prediction (SSLP) --- p.58
Chapter 4.4 --- Practical implementation issues --- p.60
Chapter 5 --- Development of a 2.0kbps speech coder --- p.67
Chapter 5.1 --- Features of the coder --- p.67
Chapter 5.2 --- Postfiltering --- p.75
Chapter 5.3 --- Voice activity detection(VAD) --- p.76
Chapter 5.4 --- Performance evaluation --- p.79
Chapter 6 --- Conclusion --- p.85
Chapter A --- Subroutine for pitch detection algorithm --- p.88
Chapter B --- Subroutines for Pitch Cycle Waveform(PCW) generation --- p.96
Chapter B.1 --- The main subroutine --- p.96
Chapter B.2 --- Subroutine for peak picking algorithm --- p.98
Chapter B.3 --- Subroutine for encoding the residue (using VDVQ) --- p.99
Chapter B.4 --- Subroutine for synthesizing PCW from its residue --- p.100
Bibliography --- p.101