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1
Bergman, Erik. "Pressure Control using Sensorless Voice Coil." Thesis, Linköpings universitet, Reglerteknik, 2013. http://urn.kb.se/resolve?urn=urn:nbn:se:liu:diva-95998.
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In this master thesis, a new method for estimating the position of the moving parts of avoice coil is presented. Instead of using a position sensor the method exploits the connectionbetween the position and the inductance of the voice coil. This is done by superpositioning a small sine voltage signal and the voice coil voltage control signal. By measuring thevoltage and current and using the fourier transform, the impedance and phase difference iscalculated which are used to compute the inductance. A medical ventilator (also known as a respirator) concept is developed with a control systemwhich takes the expiratory pressure from a higher to a lower level. The position estimationalgorithm is then used in an attempt to improve the pressure control. The result is a slightlymore stable control system. The master thesis is conducted at Maquet Critical Care (MCC) in Solna, Sweden. MCC is amedical technology company working with high performance medical ventilators. The longterm goal of this work is to develop a ventilator which is more comfortable for the patient.
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Ram, Abhishek. "Assessment of Voice Over IP as a solution for Voice over ADSL." Thesis, Virginia Tech, 2002. http://hdl.handle.net/10919/33135.
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Voice over DSL (VoDSL) is a technology that enables the transport of data and multiple voice calls over a single copper-pair. VoDSL employs packet voice technology instead of the traditional circuit switched voice. Voice over ATM (VoATM) and Voice over IP (VoIP) are the two main alternatives for carrying voice packets over DSL. ATM is currently the preferred technology, since it offers the advantage of ATMâ s built-in Quality of Service (QoS) mechanisms. IP, on the other hand, cannot provide QoS guarantees in its traditional form. IP QoS mechanisms have been evolved only in the recent years. VoIP has gained popularity in the core networks. If it could replace VoATM in the access networks, it would open the door for end-to-end IP telephony that would result in major cost savings. In this thesis, we propose a VoIP-based VoDSL architecture that provides QoS guarantees comparable to those offered by ATM in the DSL access network.
Our QoS architecture supports Premium and Regular service categories for voice traffic and the Best-Effort service category for data traffic. Voice and data packets are placed in separate output queues at the bottleneck link. The Weighted Fair Queuing algorithm in used to schedule voice and data packets for transmission over the bottleneck link. Fragmentation of large data packets reduces the waiting time for voice packets in the link. We also propose a new admission control mechanism called Admission Control by Implicit Signaling. This mechanism takes advantage of application layer signaling by mapping it to the IP header. The router can infer the resource requirements for the connection by looking at certain field in the IP header of the application layer signaling packets. This eliminates the need for an explicit signaling protocol.
We evaluate the performance of our QoS architecture by means of a simulation study. Our primary metrics are the end-to-end delay of voice packets across the access network and the bandwidth consumed by a voice call. Our results show that the end-to-end delays of voice packets in our VoIP architecture are comparable to that in the VoATM architecture. ACIS limits the number of voice calls admitted into the premium service class and provides guaranteed service to those calls under all loads. It also provides acceptable service to regular calls under light loads. We also show that PPP is a better choice than ATM as a Layer 2 protocol for our VoIP architecture. PPP offers the advantages of low bandwidth requirement and interleaving of voice packets in between fragments of large data packets during transmission over the bottleneck link. We conclude that our VoIP architecture would be suitable for future VoDSL deployments.Master of Science
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Ansari, Rukhsana 1971 Carleton University Dissertation Engineering Systems and Computer. "Compressed voice in integrated services frame relay networks: voice synchronization and congestion control." Ottawa.:, 1995.
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Wiselyn, Jeyapaul Ebby. "GSM Voice Mail Service TDM Call Control." Thesis, Uppsala universitet, Institutionen för informationsteknologi, 2012. http://urn.kb.se/resolve?urn=urn:nbn:se:uu:diva-189242.
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The Voice Mail Service (VMS) enables forwarding of calls to a dedicated Voice Mail Server (VMS) on behalf of the call receiving subscriber during certain conditions such as 'busy subscriber', 'no answer', 'always', etc. The standardization forum 3GPP has specified the Global System for Mobile communication (GSM) while the standardization forum ITU-T has specified the Integrated Services Digital Networks (ISDN) User Part (ISUP) call control protocol. Both of these standards rely on the use of Time Division Multiplex (TDM) as a media bearer and SS7 as signalling bearer, where both bearers require use of very expensive telecom-specific hardware. The thesis proposes the solution to use RTP as media bearer and IP as signalling bearer towards the handset in GSM and only use TDM as media bearer and SS7 as signalling bearer towards the VMS. The thesis demonstrates the feasibility and the advantages provided, by creating an implementation in Erlang/OTP and testing it to check if it confirms to the specification.
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Thibault, François. "High-level control of singing voice timbre transformations." Thesis, McGill University, 2004. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=81514.
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The sustained increase in computing performance over the last decades has brought enough computing power to perform significant audio processing in affordable personal computers. Following this revolution, we have witnessed a series of improvements in sound transformation techniques and the introduction of numerous digital audio effects to modify effectively the time, pitch, and loudness dimensions of audio signals. Due to the complex and multi-dimensional nature of timbre however, it is significantly more difficult to achieve meaningful and convincing qualitative transformations. The tools currently available for timbre modifications (e.g. equalizers) do not operate along perceptually meaningful axes of singing voice timbre (e.g. breathiness, roughness, etc.) resulting in a transformation control problem. One of the goals of this work is to examine more intuitive procedures to achieve high-fidelity qualitative transformations explicitly controlling certain dimensions of singing voice timbre. Quantitative measurements (i.e. voice timbre descriptors) are introduced and used as high-level controls in an adaptive processing system dependent on the characteristics observed in the input signal.The transformation methods use a harmonic plus noise representation from which voice timbre descriptors are derived. This higher-level representation, closer to our perception of voice timbre, offers more intuitive controls over timbre transformations. The topics of parametric voice modeling and timbre descriptor computation are first introduced, followed by a study of the acoustical impacts of voice breathiness variations. A timbre transformation system operating specifically on the singing voice quality is then introduced with accompanying software implementations, including an example digital audio effect for the control and modification of the breathiness quality on normal voices.
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Stepp, Cara Elizabeth. "Electromyographic control of prosthetic voice after total laryngectomy." Thesis, Massachusetts Institute of Technology, 2008. http://hdl.handle.net/1721.1/45857.
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Thesis (S.M.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 2008.Includes bibliographical references (p. 41-43).
The electrolarynx (EL) is a common rehabilitative speech aid for individuals who have undergone laryngectomy, but typical devices lack pitch control and require the exclusive use of one hand. This study investigated the viability of surface electromyography (sEMG) to control the onset and offset of an EMG-controlled EL (EMG-EL) while attending to real-time sEMG biofeedback using sEMG collected from seven locations across the ventral neck and face surface in eight individuals at least 1 year past total laryngectomy.Speech performance was assessed as the percentage of fully voiced words and successfully produced pauses. During use of the EMG-EL with biofeedback participants increased the sEMG during words and decreased the sEMG during pauses. Electrodes on the superior ventral neck, submental surface, and below the comer of the mouth showed consistently high performance across all participants. These results indicate promise for the applicability of the EMG-EL across a large segment of the laryngectomy population.
by Cara Elizabeth Stepp.
S.M.
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Liu, Purong. "Voice Control of Fetch Robot Using Amazon Alexa." Thesis, Virginia Tech, 2020. http://hdl.handle.net/10919/97439.
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With the rapid development of computers and technology, virtual assistants (VA) are becoming more and more common and intelligent. However, virtual assistants, such as Apple's Siri, Amazon's Alexa, and Google Assistant, do not currently have any physical functions. As an important part of the internet of things (IoT), the field of robotics has become a new trend in the usage of VA. In this project, a mobile robot, Fetch, is connected with the Amazon Echo Dot through the Amazon web service (AWS) and a local robot operation system (ROS) bridge server. We demonstrated that the robot could be controlled by voice commands through an Amazon Alexa. Given certain commands, Fetch was able to move in a desired direction as well as track and follow a target object. The follow model was also learned by Neural Network training, which allows for the target position to be predicted in future maps.Master of Science
Nowadays, virtual personalized assistants (VPAs) exist everywhere around us. For example, Siri or android VPAs exist on every smartphone. More and more people are getting household Virtual Assistants, such as Amazon Alexa, Google Assistant, and Microsoft's Cortana. If the virtual assistants can connect with objects which have physical functions like an actual robot, they will be able to provide better services and more functions for humans. In this project, a mobile robot, Fetch, is connected with the Echo dot from Amazon. This connection allows us to control the robot by voice command. You can ask the robot to move in a given direction or track and follow a certain object. In order to let the robot learn how to predict the position of the target when the target is lost, a map is built as an influence factor. Since a designed algorithm of target position prediction is difficult to implement, we opted to use a machine learning method instead. Therefore, a machine learning algorithm was tested on the following model.
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Höijer, David, and Hannes Jansson. "Voice-controlled order system." Thesis, Högskolan i Halmstad, Akademin för informationsteknologi, 2021. http://urn.kb.se/resolve?urn=urn:nbn:se:hh:diva-45033.
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To order pick-up food by using your computer or phone is nothing new. Food delivery companies such as FoodHero and Uber Eats along with many other around the world base their entire company idea around the food order and delivery process. For a company to stand out in such a vast market can sometimes be quite tricky. Sometimes your company needs a niche to stand out in the crowd. This project aims to create such a niche in an order system prototype based on voice-controlled systems and conversation. This prototype allows users to place food orders through only the use of natural speech and a voice assistant. The prototype utilizes products and services from both Amazon and Google to create the order system structure. The ordering system also takes advantage of the serverless architecture that both Amazon and Google provide. The end result of this project is a simple, convenient, and user-friendly prototype
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Calitz, Wietsche Roets. "Independent formant and pitch control applied to singing voice." Thesis, Stellenbosch : University of Stellenbosch, 2004. http://hdl.handle.net/10019.1/16267.
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Thesis (MScIng)--University of Stellenbosch, 2004.ENGLISH ABSTRACT: A singing voice can be manipulated artificially by means of a digital computer for the purposes of creating new melodies or to correct existing ones. When the fundamental frequency of an audio signal that represents a human voice is changed by simple algorithms, the formants of the voice tend to move to new frequency locations, making it sound unnatural. The main purpose is to design a technique by which the pitch and formants of a singing voice can be controlled independently.
AFRIKAANSE OPSOMMING: Onafhanklike formant- en toonhoogte beheer toegepas op ’n sangstem: ’n Sangstem kan deur ’n digitale rekenaar gemanipuleer word om nuwe melodie¨e te skep, of om bestaandes te verbeter. Wanneer die fundamentele frekwensie van ’n klanksein (wat ’n menslike stem voorstel) deur ’n eenvoudige algoritme verander word, skuif die oorspronklike formante na nuwe frekwensie gebiede. Dit veroorsaak dat die resultaat onnatuurlik klink. Die hoof oogmerk is om ’n tegniek te ontwerp wat die toonhoogte en die formante van ’n sangstem apart kan beheer.
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Anderson, Monty J. "Active Control of the Human Voice from a Sphere." BYU ScholarsArchive, 2015. https://scholarsarchive.byu.edu/etd/5295.
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This work investigates the application of active noise control (ANC) to speech. ANC has had success reducing tonal noise. In this work, that success was extended to noise that is not completely tonal but has some tonal elements such as speech. Limitations such as causality were established on the active control of human speech. An optimal configuration for control actuators was developed for a sphere using a genetic algorithm. The optimal error sensor location was found from exploring the nulls associated with the magnitude of the radiated pressure with reference to the primary pressure field. Both numerically predicted and experimentally validated results for the attenuation of single frequency tones were shown. The differences between the numerically predicted results for attenuation with a sphere present in the pressure field and monopoles in the free-field are also discussed.The attenuation from ANC of both monotone and natural speech is shown and a discussion about the effect of causality on the results is given. The sentence “Joe took father’s shoe bench out” was used for both monotone and natural speech. Over this entire monotone speech sentence, the average attenuation was 8.6 dB with a peak attenuation of 10.6 dB for the syllable “Joe”. Natural speech attenuation was 1.1 dB for the sentence average with a peak attenuation on the syllable “bench” of 2.4 dB. In addition to the lower attenuation values for natural speech, the pressure level for the word “took” was increased by 2.3 dB. Also, the harmonic at 420 Hz in the word “father’s” of monotone speech was reduced globally up to 20 dB. Based on the results of the attenuation of monotone and natural speech, it was concluded that a reasonable amount of attenuation could be achieved on natural speech if its correlation could approach that of monotone speech.
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Forren, Lynda Michelle Gray. "A comparison of voice-augmented and keyboard control in a supervisory control task." Thesis, Georgia Institute of Technology, 1986. http://hdl.handle.net/1853/29825.
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Brännström, Nils. "Voice-over-IP over Enhanced Uplink." Thesis, Linköping University, Department of Electrical Engineering, 2007. http://urn.kb.se/resolve?urn=urn:nbn:se:liu:diva-8479.
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The traditional voice service in mobile networks is an important service that mobile users expect high quality from. With the convergence of mobile networks towards an all-IP network, an IP-based speech service becomes important which is referred to as Voice-over-IP (VoIP). The traditional voice service is highly optimized and a VoIP service must therefore fulfil strict quality requirements to provide the same speech service quality. The air interface technology, WCDMA, which is used in third generation communication systems in Europe is constantly developed. An improved concept for the mobile-to-network transmission, called the Enhanced Uplink (EUL) provides for higher uplink capacity for packet data services. It also includes features that may provide a sufficient VoIP service quality in mobile networks, when considering the uplink transmission.
The purpose of this thesis is to evaluate the VoIP capacity over EUL and identify crucial aspects of radio resource management in order to increase the capacity. This is done through dynamic system simulations, using a realistic VoIP traffic model. The VoIP capacity is also estimated by a derived theoretical framework.\newline
It is shown by simulation results and theoretical estimations, that power control is a vital mechanism in order to increase the capacity. Simulation results indicate that a VoIP over EUL capacity of 65\% of the traditional voice service capacity may be reached. The results also indicate that to improve the capacity for larger cells, the allowed VoIP packet delay must be increased.
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Thrall, Robert Eugene. "The effect of locus of control and type of voice on satisfaction with voice and procedural justice." CSUSB ScholarWorks, 2002. https://scholarworks.lib.csusb.edu/etd-project/2058.
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This study examined the roles of type of voice and locus of control on satisfaction with type of voice and on feelings of procedural justice. Two forms of voice were assessed, instrumental and non-instrumental, as well as two forms of locus of control, external and internal. Participants read a scenario that randomly placed them into type of voice. Participants responded to surveys to determine the persons' locus of control, satisfaction with voice and feelings of procedural justice. A form of participation that brings employees satisfaction is voice. Allowing employees to express their opinions is seen as fair and has benefits to employees, as well as the organization. Some individuals prefer to have an impact and be more involved in the workplace, while others do not.
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Lambertsson, Christoffer. "Expectations of Privacy in Voice Interaction – A Look at Voice Controlled Bank Transactions." Thesis, KTH, Skolan för datavetenskap och kommunikation (CSC), 2017. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-207155.
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There is a strong mainstream interest in the tech world for voice interaction, in parts thanks to the technological advancements in the field and the use of big data which in some cases make speech recognition on par with humans. The aim of this paper is to explore the relationship between voice interaction and privacy. Two prototypes on voice interaction of bank services were created, they had differences in feedback, correctness and need of physical interaction. Twelve participants were instructed to perform tasks like checking balance and transferring money with the prototypes. Then they were asked about their relation to privacy in the context of their previous experiences with voice interaction and the two prototypes they used. The results show that what was perceived as private and the need for privacy varied a lot between the participants. In the end a respectful and empowering design to voice interaction is proposed in which the user could customize how the application provides feedback by adding contextual sensitivity e.g. using GPS to know if you are home, or a camera to know if you are looking at the device.Det finns ett starkt intresse inom IT-branschen för röstinteraktion, delvis tack vare de teknologiska framstegen inom området, och användandet av stora mängder data vilket i vissa fall gör röstigenkänning likställd med människor. Målet med denna rapport är att utforska förhållandet mellan röstinteraktion och personlig integritet. Två prototyper för röstinteraktion av banktjänster skapades, de hade olika återkoppling, korrekthet och behov av fysisk interaktion. Tolv deltagare instruerades att utföra uppgifter såsom att kontrollera kontobalansen och överföra pengar med hjälp av prototyperna. De svarade sedan på frågor om deras relation till personlig integritet i sammanhanget av deras tidigare upplevelser med röstinteraktion och de två prototyper de provat. Resultatet visar att vad som upplevs som privat, och behovet av personlig integritet, varierade mycket mellan deltagarna. I slutändan föreslås en respekterande och stärkande design av röstinteraktion i vilken användaren kan skräddarsy hur applikationen ger återkoppling genom att lägga till kontextuell känslighet t.ex. genom GPS för att veta om du är hemma, eller en kamera för att veta om du tittar på enheten.
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Ardaillon, Luc. "Synthesis and expressive transformation of singing voice." Thesis, Paris 6, 2017. http://www.theses.fr/2017PA066511/document.
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Le but de cette thèse était de conduire des recherches sur la synthèse et transformation expressive de voix chantée, en vue de pouvoir développer un synthétiseur de haute qualité capable de générer automatiquement un chant naturel et expressif à partir d’une partition et d’un texte donnés. 3 directions de recherches principales peuvent être identifiées: les méthodes de modélisation du signal afin de générer automatiquement une voix intelligible et naturelle à partir d’un texte donné; le contrôle de la synthèse, afin de produire une interprétation d’une partition donnée tout en transmettant une certaine expressivité liée à un style de chant spécifique; la transformation du signal vocal afin de le rendre plus naturel et plus expressif, en faisant varier le timbre en adéquation avec la hauteur, l’intensité et la qualité vocale. Cette thèse apporte diverses contributions dans chacune de ces 3 directions. Tout d’abord, un système de synthèse complet a été développé, basé sur la concaténation de diphones. L’architecture modulaire de ce système permet d’intégrer et de comparer différent modèles de signaux. Ensuite, la question du contrôle est abordée, comprenant la génération automatique de la f0, de l’intensité, et des durées des phonèmes. La modélisation de styles de chant spécifiques a également été abordée par l’apprentissage des variations expressives des paramètres de contrôle modélisés à partir d’enregistrements commerciaux de chanteurs célèbres. Enfin, des investigations sur des transformations expressives du timbre liées à l'intensité et à la raucité vocale ont été menées, en vue d'une intégration future dans notre synthétiseurThis thesis aimed at conducting research on the synthesis and expressive transformations of the singing voice, towards the development of a high-quality synthesizer that can generate a natural and expressive singing voice automatically from a given score and lyrics. Mainly 3 research directions can be identified: the methods for modelling the voice signal to automatically generate an intelligible and natural-sounding voice according to the given lyrics; the control of the synthesis to render an adequate interpretation of a given score while conveying some expressivity related to a specific singing style; the transformation of the voice signal to improve its naturalness and add expressivity by varying the timbre adequately according to the pitch, intensity and voice quality. This thesis provides some contributions in each of those 3 directions. First, a fully-functional synthesis system has been developed, based on diphones concatenations. The modular architecture of this system allows to integrate and compare different signal modeling approaches. Then, the question of the control is addressed, encompassing the automatic generation of the f0, intensity, and phonemes durations. The modeling of specific singing styles has also been addressed by learning the expressive variations of the modeled control parameters on commercial recordings of famous French singers. Finally, some investigations on expressive timbre transformations have been conducted, for a future integration into our synthesizer. This mainly concerns methods related to intensity transformation, considering the effects of both the glottal source and vocal tract, and the modeling of vocal roughness
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D'Ippolito, Tommaso. "Voice control in a graphical user interface environment, human factors implications." Thesis, National Library of Canada = Bibliothèque nationale du Canada, 1997. http://www.collectionscanada.ca/obj/s4/f2/dsk2/ftp04/mq22072.pdf.
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D'Ippolito, Tommaso Carleton University Dissertation Psychology. "Voice control in a graphical user interface environment; human factors implications." Ottawa, 1997.
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Khan, Mohd Tauheed. "Multimodal Data Fusion Using Voice and Electromyography Data for Robotic Control." University of Toledo / OhioLINK, 2019. http://rave.ohiolink.edu/etdc/view?acc_num=toledo156440368925597.
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Pappas, Johnny J. "A Flexible Voice Communication System for a Real-Time Mission Control Facility." International Foundation for Telemetering, 1992. http://hdl.handle.net/10150/611925.
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International Telemetering Conference Proceedings / October 26-29, 1992 / Town and Country Hotel and Convention Center, San Diego, CaliforniaDue to the complexity of real-time missions, an increasing number of participants, and the critical nature of test missions, providing a reliable, versatile voice communication network for mission support entities has become essential. A voice communication system has a direct impact on the effectiveness of every mission and the safety of mission personnel. Each participant must satisfy unique functional and operational communication requirements. This paper addresses the functional, operational, and ergonomic aspects associated with a voice communication system for the Central Control Facility (CCF) at the Air Force Development Test Center (AFDTC), Eglin AFB, Florida. The communication system was purchased from an Edwards AFB Digital Switch requirements contract.
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Päärni, Anna. "Designing a Voice Controlled Interface For Radio : Guidelines for The First Generation of Voice Controlled Public Radio." Thesis, Umeå universitet, Institutionen för tillämpad fysik och elektronik, 2017. http://urn.kb.se/resolve?urn=urn:nbn:se:umu:diva-136894.
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From being a fictional element in sci-fi, voice control has become a reality, with inventions such as Apple's Siri, and interactive voice response (IVR) when calling your doctor's office. The combination of radio’s strength as a hands-free medium, public radio’s mission to reach across all platforms and the rise of voice makes up a relevant intersection; voice controlled public radio in Sweden. This thesis has aimed to investigate how radio listeners wish to interact using voice control to listen to on-demand radio content. Additionally, what does public radio need to consider when implementing the UX for voice control? A questionnaire, Google Analytics-data from Sveriges Radio’s mobile app and website, and an interview with National Public Radio were used to determine four initial functions voice controlled on-demand. The functions were turned into four scenarios, used as the basis for a workshop and brainstorming session. Three of these scenarios were then selected to be the foundation for a Wizard of Oz-prototype, which six users evaluated. The results indicate that to create a viable voice user interface (VUI) for radio, there needs to be a profile for the user to create a personalized experience, with filtered searches created by user behaviour and preference settings. The VUI also needs to allow synonyms for program names, as well as keyword tag material for clustering similar material and to enable personalized user searches. This keyword-system can further be utilized to give user recommendations based on behaviour and preferences. A companion application with a graphical user interface (GUI) can be used for such functionality as settings, help, queuing and creating playlists. Finally, the system needs to ensure that the users are taken care of and guided. A radio host can be used as a basis for the system's persona, to create a consistent and familiar interface. Aspects to examine in the future are more extensive user testing for the VUI design, filter bubbles, ensuring that the user data is not misused or leaked, dimensions of the system's persona, the future of radio and the progress of IPAs.
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Hull, Darcey M. "Thyroarytenoid and cricothyroid muscular activity in vocal register control." Thesis, University of Iowa, 2013. https://ir.uiowa.edu/etd/4994.
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Register and pitch are two distinct perceptual entities of the human voice. Without clear evidence for the use of the terminology, sources have begun to refer to lighter, or "falsetto", register as being "cricothyroid dominant" and heavier, or "chest", register as being "thyroarytenoid dominant" (Hirano, 1987; Miller, 1996; McCoy, 2004; Henrich, 2006; Spivey, 2008; Edwin, 2008; Phillips, Williams, & Edwin, 2012). The same intrinsic laryngeal musculature (i.e. the cricothyroid, CT, and thyroarytenoid, TA, muscles) play a role in the control of both register and pitch. Higher-pitched phonation, typically used to produce falsetto register, is mediated primarily by the cricothyroid (CT) muscle. The thyroarytenoid (TA) muscle plays a larger role in controlling lower-pitched voicing, the pitch range in which chest register tends to be used (Titze, 1989b; Shipp and McGlone, 1971). Despite their frequent co-occurrence, high and low pitched phonation are not controlled in the same way as light and heavy register productions.
The purpose of this study was to examine the ratio of CT and TA muscular activity in the control of chest and falsetto registers. Data were collected from untrained voice users: four females and one male. Hooked-wire electrodes were inserted into both the CT and TA muscles of each participant in order to collect electromyographic (EMG) data during glissando from low to high pitch on the vowel /i/ twice per subject, and tasks eliciting maximal activation of CT and TA muscles. A trained singing instructor with 17 years of experience determined and recorded the occurrence of register transition during each glissando. CT and TA EMG activity data from the glissando were normalized relative to maximum elicited CT and TA EMG activity, and were then retrospectively analyzed. CT muscular dominance was defined as a ratio of percentage of maximum CT EMG activity to percentage of maximum TA EMG activity greater than 1 (i.e. CT:TA greater than 1). TA dominance is a ratio of CT:TA activity less than 1 (i.e. CT:TA less than 1).
During glissando, all subjects experienced register transition from chest to falsetto register. In all subjects, the majority of chest register, and all of falsetto register, was produced with CT muscular dominance. Only the 3-4 lowest semitones, on average, in chest register were TA dominant. The transition from chest to falsetto register consistently did occur when the CT muscle was dominant, however, register transition did not occur as CT muscle activity began to dominate TA muscle activity. Results of the study showed that CT muscular dominance did not define falsetto register, nor was chest register defined by the TA muscular dominance.
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Stowell, Dan. "Making music through real-time voice timbre analysis : machine learning and timbral control." Thesis, Queen Mary, University of London, 2010. http://qmro.qmul.ac.uk/xmlui/handle/123456789/412.
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People can achieve rich musical expression through vocal sound { see for example human beatboxing, which achieves a wide timbral variety through a range of extended techniques. Yet the vocal modality is under-exploited as a controller for music systems. If we can analyse a vocal performance suitably in real time, then this information could be used to create voice-based interfaces with the potential for intuitive and ful lling levels of expressive control. Conversely, many modern techniques for music synthesis do not imply any particular interface. Should a given parameter be controlled via a MIDI keyboard, or a slider/fader, or a rotary dial? Automatic vocal analysis could provide a fruitful basis for expressive interfaces to such electronic musical instruments. The principal questions in applying vocal-based control are how to extract musically meaningful information from the voice signal in real time, and how to convert that information suitably into control data. In this thesis we address these questions, with a focus on timbral control, and in particular we develop approaches that can be used with a wide variety of musical instruments by applying machine learning techniques to automatically derive the mappings between expressive audio input and control output. The vocal audio signal is construed to include a broad range of expression, in particular encompassing the extended techniques used in human beatboxing. The central contribution of this work is the application of supervised and unsupervised machine learning techniques to automatically map vocal timbre to synthesiser timbre and controls. Component contributions include a delayed decision-making strategy for low-latency sound classi cation, a regression-tree method to learn associations between regions of two unlabelled datasets, a fast estimator of multidimensional di erential entropy and a qualitative method for evaluating musical interfaces based on discourse analysis.
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Russell, Katherine M. "Hysterectomy, Metaphor, and Voice: An Exploratory Study of Surgery Experiences." Antioch University / OhioLINK, 2017. http://rave.ohiolink.edu/etdc/view?acc_num=antioch1492009348790118.
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Raman, Sujatha. "The relevance of STS to peach research : the need for a third voice on nuclear strategy /." Thesis, This resource online, 1991. http://scholar.lib.vt.edu/theses/available/etd-10102009-020151/.
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McBean, John M. (John Michael) 1979. "Design and control of a voice coil actuated robot arm for human-robot interaction." Thesis, Massachusetts Institute of Technology, 2004. http://hdl.handle.net/1721.1/17951.
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Thesis (S.M.)--Massachusetts Institute of Technology, Dept. of Mechanical Engineering, 2004.Includes bibliographical references (leaf 68).
The growing field of human-robot interaction (HRI) demands robots that move fluidly, gracefully, compliantly and safely. This thesis describes recent work in the design and evaluation of long-travel voice coil actuators (VCAs) for use in robots intended for interacting with people. The basic advantages and shortcomings of electromagnetic actuators are discussed and evaluated in the context of human-robot interaction, and are compared to alternative actuation technologies. Voice coil actuators have been chosen for their controllability, ease of implementation, geometry, compliance, biomimetic actuation characteristics, safety, quietness, and high power density. Several VCAs were designed, constructed, and tested, and a 4 Degree of Freedom (DOF) robotic arm was built as a test platform for the actuators themselves, and the control systems used to drive them. Several control systems were developed and implemented that, when used with the actuators, enable smooth, fast, life-like motion.
by John M. McBean.
S.M.
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Zhang, Shu Carleton University Dissertation Engineering Systems and computer. "Congestion control in frame relay networks with variable bit rate compressed voice and data traffic." Ottawa, 1993.
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Al, Hashimi Sama'a. "Paralinguistic vocal control of interactive media : how untapped elements of voice might enhance the role of non-speech voice input in the user's experience of multimedia." Thesis, Middlesex University, 2007. http://eprints.mdx.ac.uk/4961/.
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Much interactive media development, especially commercial development, implies the dominance of the visual modality, with sound as a limited supporting channel. The development of multimedia technologies such as augmented reality and virtual reality has further revealed a distinct partiality to visual media. Sound, however, and particularly voice, have many aspects which have yet to be adequately investigated. Exploration of these aspects may show that sound can, in some respects, be superior to graphics in creating immersive and expressive interactive experiences. With this in mind, this thesis investigates the use of non-speech voice characteristics as a complementary input mechanism in controlling multimedia applications. It presents a number of projects that employ the paralinguistic elements of voice as input to interactive media including both screen-based and physical systems. These projects are used as a means of exploring the factors that seem likely to affect users' preferences and interaction patterns during non-speech voice control. This exploration forms the basis for an examination of potential roles for paralinguistic voice input. The research includes the conceptual and practical development of the projects and a set of evaluative studies. The work submitted for Ph.D. comprises practical projects (50 percent) and a written dissertation (50 percent). The thesis aims to advance understanding of how voice can be used both on its own and in combination with other input mechanisms in controlling multimedia applications. It offers a step forward in the attempts to integrate the paralinguistic components of voice as a complementary input mode to speech input applications in order to create a synergistic combination that might let the strengths of each mode overcome the weaknesses of the other.
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Thorell, Hampus. "Voice Activity Detection in the Tiger Platform." Thesis, Linköping University, Department of Electrical Engineering, 2006. http://urn.kb.se/resolve?urn=urn:nbn:se:liu:diva-6586.
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Sectra Communications AB has developed a terminal for encrypted communication called the Tiger platform. During voice communication delays have sometimes been experienced resulting in conversational complications.
A solution to this problem, as was proposed by Sectra, would be to introduce voice activity detection, which means a separation of speech parts and non-speech parts of the input signal, to the Tiger platform. By only transferring the speech parts to the receiver, the bandwidth needed should be dramatically decreased. A lower bandwidth needed implies that the delays slowly should disappear. The problem is then to come up with a method that manages to distinguish the speech parts from the input signal. Fortunately a lot of theory on the subject has been done and numerous voice activity methods exist today.
In this thesis the theory of voice activity detection has been studied. A review of voice activity detectors that exist on the market today followed by an evaluation of some of these was performed in order to select a suitable candidate for the Tiger platform. This evaluation would later become the foundation for the selection of a voice activity detector for implementation.
Finally, the implementation of the chosen voice activity detector, including a comfort noise generator, was done on the platform. This implementation was based on the special requirements of the platform. Tests of the implementation in office environments show that possible delays are steadily being reduced during periods of speech inactivity, while the active speech quality is preserved.
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Nguyen, Manh Cuong. "Voice capacity over LTE in PMR context : challenges and solutions." Thesis, Evry, Institut national des télécommunications, 2015. http://www.theses.fr/2015TELE0015/document.
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Le réseau de radio communications mobiles professionnelles (PMR), qui est utilisé pour le fonctionnement de la sécurité publique, doit évoluer pour les solutions à large bande pour satisfaire les demandes des utilisateurs à l’avenir. Dans les technologies à large bande actuels, Long Term Evolution (LTE), développé par le 3GPP (3rd Generation Partnership Project), est considéré comme l’un des candidats potentiels pour la prochaine génération de PMR. Malgré le fait que la technologie LTE en charge la transmission de données à haute vitesse et prend en charge différentes tailles de paquets en utilisant la modulation et le codage adaptatifs (AMC), le LTE n’est pas encore optimisée pour la communication vocale à bas débit, en particulier en cas d’utilisation LTE pour la radio communications mobiles professionnelles (PMR ) contexte. Par conséquent, dans cette thèse, nous présentons des solutions pour renforcer la capacité de la voix de la technologie LTE dans le cadre PMR à la fois la liaison montante et la transmission de liaison descendante. Les nouvelles propositions, basées sur la technologie de norme LTE existant avec des adaptations, permettent la réduction de les frais généraux de données et les frais généraux de contrôle sur LTE (VoLTE) dans le contexte PMRThe Professional Mobile Radio (PMR) network, which is used for public safety operation, has to evolve to the broadband solutions to satisfy the user demands in the future. In the current broadband technologies, Long Term Evolution (LTE) standard, developed by the 3GPP (3rd Generation Partnership Project), is considered one of the potential candidates for the next generation of PMR. Despite the fact that LTE supports high-speed data transmission and supports different packet sizes by using Adaptive Modulation and Coding (AMC), LTE is not yet optimized for low bit rate voice communication, especially in case of using LTE in Professional Mobile Radio (PMR) context. Therefore, in this dissertation, we present several new solutions for enhancing the voice capacity of LTE in the PMR context for both uplink and downlink transmission. The new propositions, based on existent LTE standard technology with adaptations, allow reducing both control and data overhead issues of Voice over LTE (VoLTE) in PMR context
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Boork, Adam, and Karin Wennersten. "Assisterande styrning av lindningsmaskiner : En förstudie om röststyrning och andra assisterande lösningar som stöd till en lindningsstation." Thesis, Uppsala universitet, Institutionen för samhällsbyggnad och industriell teknik, 2020. http://urn.kb.se/resolve?urn=urn:nbn:se:uu:diva-414366.
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Lindningsstationen på ABB Power Transformers i Ludvika är idag styrd med fotpedal, som på senare år fått stor kritik kring dess ergonomiska utformning och risk för belastningsskador. I en studie år 2011 testades ett system med röststyrning som bemöttes positivt inom det ergonomiska perspektivet, men negativt inom dess prestanda och bristande säkerhetsklassning. Strax efteråt lades konceptet på is, till vilket företaget nu villee se vart tekniken kring röststyrning och assisterande lösningar ligger idag. Syftet med denna rapport är att skapa ett jämförande underlag för vidare investeringsbeslut av olika lösningar relaterade till industriell röststyrning och assisterande lösningar, vilket inkluderar hur de kan appliceras på den nuvarande lindningsmaskinen. Genom en litteraturstudie och intervjuer skapades 1) en lista med dem kundbehov som lösningen skulle uppfylla samt en lista med kundbehov som lösningen var tvungen att uppfylla samt 2) produktkrav som en trådlös nödstoppslösning var tvungen att följa. Dessa ingångsparametrar användes sedan i en marknadsundersökning där olika företag och system kring röststyrning, assisterande lösningar och nödstopp jämfördes., där tre Tre förslag på röststyrning och två nödstopp gick vidare för analys kring konceptuell implementering. Slutsatsen av arbetet blev att tekniken kring industriell röststyrning fortfarande är i en utvecklande fas, där en lösning idag kräver en kombination av flera mindre komponenter. De nödstoppslösningar som identifierades kan dock inköpas och implementeras idag. Om företaget väljer att investera i en lösning inom en snar framtid, rekommenderas en uppgradering av den tidigare lösningens hårdvara från Honeywell. Är företaget villiga att lägga lite mer tid och resurser på lösningen rekommenderas Voice INTER Connect:s lösning. Alternativt avvakta tills ITSpeex kan erbjuda en fullständig lösning.The winding station at ABB Power Transformers in Ludvika is today controlled using a foot pedal, which in recent years has received a lot of criticism for its poor ergonomic design and risk for repetitive strain injuries. In a study performed in 2011, a voice control system was tested, and positively received from an ergonomic perspective, but negatively when it came to both performance and proper safety. Shortly afterwards, the concept was put on hold, to which the company now wants to see where the technology surrounding voice control and assistive solutions are today. The purpose of this report is to create a comparative foundation for future investment decisions, about different solutions related to industrial voice control and assistive solutions, which includes how the solutions are applied at the current winding station. Through a literature study and interviews,1) a list of customer needs was made, which the voice solution had to fulfill, and 2) product requirements which the wireless emergency stop had to follow. These were later used in a market research where different companies and systems involving voice control, assistive solutions and emergency stops were compared. Three proposals for voice control and two for emergency stop moved on to further analysis about conceptual implementation. The conclusion drawn from the study was that the industrial voice control technology is still under development, where a contemporary solution requires a combination of several smaller components. The identified emergency stops solutions can, however, be purchased and installed today. If the company chooses to invest in a solution within the near future, an upgrade of the previous solution from Honeywell is recommended. With time and resources available, a solution from Voice INTER Connect is the one to pick. Alternatively hold off until ITSpeex can offer a complete solution.
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Umbert, Morist Martí. "Expression control of singing voice synthesis: modeling pitch and dynamics with unit selection and statistical approaches." Doctoral thesis, Universitat Pompeu Fabra, 2016. http://hdl.handle.net/10803/361103.
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This thesis focuses on the control of a singing voice synthesizer to achieve natural expression similar to a real singer. There are many features that should be controlled to achieve natural expression related to melody, dynamics, rhythm, and timbre. In this thesis we focus on the control of pitch and dynamics with a unit selection-based system, two statistically-based systems, and a hybrid system. These systems are trained with two possible expression databases that we have designed, recorded, and labeled. We define the basic units from which the databases are built of, which are basically sequences of three notes or rests. Our perceptual evaluation compares the proposed systems with other systems to see how these relate to each other. The objective evaluation focuses on the algorithms efficiency.Aquesta tesi es centra en el control dels sintetitzadors de veu cantada per aconseguir una expressivitat natural semblant a la d'un cantant real. Hi ha moltes característiques que s'haurien de controlar per aconseguir una expressivitat natural relacionades amb la melodia, la dinàmica, el ritme i el timbre. En aquesta tesi ens centrem en el control de la freqüència fonamental i de la dinàmica amb un sistema basat en selecció d'unitats, dos sistemes estadístics, i un sistema híbrid. Aquests sistemes són entrenats amb dues possibles bases de dades expressives que hem dissenyat, enregistrat i etiquetat. Hem definit les unitats bàsiques a partir de les quals les bases de dades s'han construit i que són seqüències de tres notes o silencis. La nostra avaluació perceptual compara els sistemes proposats amb altres sistemes per tal de veure com els podem relacionar. L'avaluació objectiva es centra en l'eficiència dels sistemes.
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Liu, Chunlei. "Wireless network enhancements using congestion coherence, faster congestion feedback, media access control and AAL2 voice trunking /." The Ohio State University, 2001. http://rave.ohiolink.edu/etdc/view?acc_num=osu1486572165276861.
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Voroneckij, Marek. "Kompiuterio valdymas balsu." Master's thesis, Lithuanian Academic Libraries Network (LABT), 2007. http://vddb.library.lt/obj/LT-eLABa-0001:E.02~2007~D_20070816_175147-23409.
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Šiame darbe yra nagrinėjamos kompiuterio valdymo balsu galimybės. Pagrindinis dėmesys yra skiriamas kalbos atpažinimo taikymams Lietuvoje ir užsienyje. Apžvelgti pasiekimai šioje srityje. Darbe išnagrinėtos programų „Atpazinimas.exe“ ir „Speech Recognition in Windows XP“ galimybės bei trūkumai. Atlikti šių programų eksperimentiniai tyrimai ir pateikti tyrimų aprašymai. Suformuluotos išvados apie tyrimų rezultatus ir šių programų taikymo galimybes.This work analyses the peculiarity of computer control by voice. The use of speech recognition in Lithuania and abroad and the achievements in this area are mostly emphasized in this work. The advantages and disadvantages of such programs as “Atpazinimas.exe“ and “Speech Recognition in Windows XP“ were researched and the results of those examinations are conveyed. The results of those experiments are summed up and the opportunities of using the programs are presented.
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Höglund, Salomon, and Hampus Nyberg. "Röststyrning för äldreboendes sängar : Ett komplement till fysiska kontroller." Thesis, Södertörns högskola, Institutionen för naturvetenskap, miljö och teknik, 2015. http://urn.kb.se/resolve?urn=urn:nbn:se:sh:diva-29788.
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Voice control as a potential method of human-computer interaction is steadilyincreasing, most recently with watches, cars and household appliances. Theelderly should be in focus for more digital development since they aresensitive to bad design. The purpose of the study therefore intends to includethis group in a survey of how a voice control system for beds can be formedfor the residents of a retirement home. The aim was to study selected components of a system that the elderly couldinteract with using their voice to control their adjustable beds. Limitationswere conducted mainly in the technical aspects of this system by excluding theuse of a technical prototype. The survey was performed using two methods.First, residents of a retirement home were individually interviewed to confirmand strenghten the purpose of the study, that they would benefit of having avoice controlled system to complement the current handheld controller ofeach of their beds. The results of the interviews lay the foundation for thesecond part of the study: user tests of a simulated system. In these tests,selected parts of a hypothetical voice control system were evaluated forusefullness. The results indicate that participants valued simplicity over complexfunctions in a voice control system. In the feedback test, participantspreferred voice over sound as information-carrying. Thus, voice control ofbeds should be developed with a simple design that is conceptually the same,or very similar, to the participants mental model of the handheld controller.
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Fransson, Linda, and Therese Jeansson. "Biometric methods and mobile access control." Thesis, Blekinge Tekniska Högskola, Avdelningen för programvarusystem, 2004. http://urn.kb.se/resolve?urn=urn:nbn:se:bth-5023.
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Our purpose with this thesis was to find biometric methods that can be used in access control of mobile access. The access control has two parts. Firstly, to validate the identity of the caller and, secondly, to ensure the validated user is not changed during the session that follows. Any solution to the access control problem is not available today, which means that anyone can get access to the mobile phone and the Internet. Therefore we have researched after a solution that can solve this problem but also on how to secure that no one else can take over an already validated session. We began to search for biometric methods that are available today to find them that would be best suited together with a mobile phone. After we had read information about them we did choose three methods for further investigation. These methods were Fingerprint Recognition, Iris Scan and Speaker Verification. Iris Scan is the method that is best suited to solve the authentication problem. The reasons for this are many. One of them is the uniqueness and stability of the iris, not even identical twins or the pair of the same individual has the same iris minutiae. The iris is also very protected behind eyelids, cornea and the aqueous humor and therefore difficult to damage. When it comes to the method itself, is it one of the most secure methods available today. One of the reasons for this is that the equal error rate is better than one in a million. However, this rate can be even better. It all depends on the Hamming Distance, which is a value that show how different the saved and temporarily template are, and what it is set to. To solve our session authentication, which was to make sure that no one else could take over a connected mobile phone, a sensor plate is the answer. This sensor will be able to sense for touch, heat and pulse. These three sensor measurements will together secure a validated session since the mobile phone will disconnect if the sensor looses its sensor data. There are, however, technological and other challenges to be solved before our proposed solutions will become viable. We address some of these issues in our thesis.
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Morris, Zackery David. "A SINGER’S STRESS: YOGA AND MEDITATION TECHNIQUES IN THE COLLEGIATE VOICE STUDIO." UKnowledge, 2019. https://uknowledge.uky.edu/music_etds/138.
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Often neglected in voice study is the understanding that a singer’s instrument lives in his/her body and therefore cannot be packed away like other musical objects. Yoga and mindfulness compliment the belief of a whole body instrument. Data gathered on college campuses report that stress and anxiety are on the rise, thus reinforcing the need for MBSR and yoga as beneficial and proven tools for stress relief. The current state of research in the realm of mental health in colleges merits the study of a cohesive layout of these exercises and their expected outcomes in singing as well as stress management. Therefore, this document will present yoga sequences that align with the core aspects of singing including breath support, fluid vocal production and artistic expression. Exercises founded on principles of mindfulness are provided to bring awareness of mental qualities within a singer. Mental qualities gained from meditation practice include improved self-esteem, lowered anxiety, and increased focus. Yoga sequences will focus on certain areas of the body commonly addressed in voice studies. Collections of yoga asana, or postures, are featured to allow singers to recognize tightness and inefficiencies in their bodies, thus improving vocal function. This guide will combine yoga traditions along with mindfulness research to introduce ancient philosophies to singers and their teachers, resulting in meaningful and productive voice lessons.
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Lu, Chun-Hung, and 呂俊鴻. "Intelligent Voice-Control Power Wheelchair System." Thesis, 2014. http://ndltd.ncl.edu.tw/handle/62318715199506976930.
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碩士德明財經科技大學
資訊管理系
102
Nowadays, the population of the elderly grows fast in Taiwan. The power wheelchair thus becomes necessary equipment for the elderly. Because of the illness and weakness, some of the elderly have difficulty to control the wheelchair by hand. Besides the elderly, some of the disabled need the power wheelchair to move around in their daily lives. Therefore, a power wheelchair system with voice-controlled function is proposed in this thesis to help those who have problem to control the power wheelchair by hand instead of controlling the power wheel chair by voice commands sent from microphone via Bluetooth system. The voice-controlled model not only uses voice to control power wheelchair direction, but also can integrate with automatic navigation model to go right spot by setting path previouly. Moreover, the system provides button control model and gyro control model which can be provided for anyone to control power wheelchair. The Wi-Fi wireless network system and Android system based smartphones are another two major parts of the proposed scheme. The automatic alert system has been also developed to call for help and GPS position by the smartphone when any dangerous situation occurred. The proposed control system can help the elder users of the power wheelchair to have a better quality of life.
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Li, Zong-Han, and 李宗翰. "Voice Assistant for Robotic Control Application." Thesis, 2018. http://ndltd.ncl.edu.tw/handle/d8q9q3.
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碩士國立勤益科技大學
資訊工程系
106
The trend towards fewer children and reducing environmental pollution, the research in artificial intelligence (AI) and industry 4.0 has become the future trend. Many industrial countries have begun to integrate various technologies and triggered a new industrial revolution. Our life has also begun into smart, unmanned, and informative. For instance, natural language processing (NLP) applications occupy a large part of the AI market. From the past decades, users have to communicate with the computer or mobile phone interface through the screen and the keyboard. However, with the introduction of intelligent voice assistants into the human environment, people can interact with voice virtual assistants through simple instructions without the need of touch. Thus, the communication between human and devices will be similar to a human-to-human dialogue. If this technology can be achieve to the industrial field, even though non-relevant background personnel can easily complete the machinery operations (of the machine) and to solve the erroneous operations caused by the lack of professional knowledge and background of the on-site personnel. Also this technology can achieve self-correction, moving items, or Real-time data query, and those requirements have become the requirements of market and those are indispensable for the future of intelligent factories in the future of intelligent factories. This study utilizes Google speech recognition engine to transform voice messages through voice device into texts. Then, the above-mentioned aforementioned methods are used employed to perform word segmentation, part-of-speech tagging and propose an improved rapid automatic keyword extraction method to correspond to the relevant execution function. In this method, the user can operate and manage any machinery various machines by means of voice interaction.
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LIN, CHING-HSIANG, and 林清祥. "Application of Voice Control in Excavator Design." Thesis, 2017. http://ndltd.ncl.edu.tw/handle/8pj495.
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碩士建國科技大學
自動化工程系暨機電光系統研究所
105
The purpose of this system is to apply the voice function to improve the traditional excavator driving system, so that the excavator to combine the voice control signal products to enhance the safety of the work area of the construction safety and convenience, the system uses voice-driven Earthwork, perform mining tasks. Voice control using voice enhancement technology to improve voice quality, improve voice clarity, in order to improve the system recognition rate and anti-jamming capability, can accurately drive the excavator action. Speech recognition is applied to the design of humanized excavator, so that the controller can follow the voice of their own voice command to do the drive demand, the system will be integrated into the excavator voice recorder, according to the specific language of the use and command Dig down, up to mention, forward, back, left, right-handed six control. The experimental design of the system can be improved to improve the traditional function of the shortcomings, and the real combination of voice function in the excavator application design purposes. Keywords: Voice Function, Excavator, Voice Enhancement, Speech Recognition Control
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Lin, Yu-Wei, and 林育暐. "A Voice Control System for Smart Home Applications." Thesis, 2017. http://ndltd.ncl.edu.tw/handle/f5apqj.
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碩士國立臺灣科技大學
電機工程系
105
As technology advances, the number of devices with networking capabilities increased in a very fast speed, and it has been gradually moving towards the age of Internet of things. As of Internet of things’ applications in human life, Smart Home system is the most distinctive representative. Companies such as Nest, Samsung develope smart home systems and smart home apps. Users can use these apps to read smart home devices’ status at any time; also user can set the status of Smart Home device through this app. However, in the progress of science and technology, human-machine interface control is no longer confined to touch control, but evolves to a new human-machine interface control ─ Voice Control. This study gives more information about how to build a voice control system with Amazon Alexa voice development technology. Besides, Google Home smart home assistant, recently released and implemented by its open development technology platform, enables developers to develop their own voice services. Finally, the research gives some analysis and result from the implementation of the two systems, and provides developers more in-depth understanding. This study is divided into two parts. The first part is the entire architecture of the systems (Amazon Alexa and Actions on Google) which contain devices, endpoints and web services. The second part is that the system mentioned above combined with IP cameras, and it can achieve a function that controls multiple cameras with voice control. In conclusion, this research implements two kinds of voice control systems of smart home, and controls multiples IP cameras. Furthermore, this paper provides the analysis between Amazon Alexa and Action on Google completely.
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Lin, Long-Bang, and 林隆邦. "Voice Assistant for Applications of Machine Tool Control." Thesis, 2018. http://ndltd.ncl.edu.tw/handle/384vtd.
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碩士國立勤益科技大學
資訊工程系
106
As advances technology, Artificial Intelligence has become a important topic in the experts system. The automatic speech recognition system have been widely used in all aspects of life. The rapid advancement and development on science and technology have had a great impact on human society. Although speech recognition technology has advanced by leaps and bounds in recent years, there are many areas that have yet to be breakthrough. This paper proposes a method for the user to control the machine tool through the instruction of the voice assistant system. The methods converts a voice signal into a text message and outputs it. In order to allow the machine to understand the commands issued by the user. This study use natural language processing to achieve semantic analysis. First, the process put the built-up corpora as the natural language processing input segmentation words and sentences. Second, the results of word segmentation are used for part-of-speech tagging and syntactic analysis. In the final step is extracted important keywords as instructions for operating the machine tool. The extracted keywords process need to be converted into machine code to correspond with the machine and execute related instructions.
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Nageshar, Nikesh. "Voice quality control in packet switched wireless networks." Thesis, 2013. http://hdl.handle.net/10539/12855.
Full textAbstract:
Wireless systems have engaged the evolutionary migration from traditional circuit switch
technology to packet based technology. Presently all next generation wireless networks have
been specified with a packet based Radio Access Networks (RAN), which implies that all the
flaws of traditional packet based networks now also apply to voice. These flaws result in
decreased speech quality derived from increased latency, jitter and packet loss. This thesis
provides the basis for a solution that will facilitate voice quality control in a packet switched
wireless network based on the integrated approach of providing Quality of Service (QoS) control
across the Admission Control (AC) component, Bearer or Service Flow component and
mapping across these components to the appropriate Quality of Service (QoS) metrics at the
transport network.
The original contribution of knowledge to the field of electrical and information engineering is
the proposal of a Quality of Service (QoS) framework and control mechanisms that result in the
transmission of quality voice over a packet switched wireless network autonomous to voice
specific signalling or media protocols. These proposals include: Heuristic Analysis in the
Admission Control (AC) component; the addition of a voice service class Admission Control
(AC) model; selection of a voice specific Bearer or Service Flow and the mapping thereof to a
voice specific Quality of Service (QoS) queue or Service Flow at the transport or backhaul
network. All these solutions are presented with the goal of ensuring the preservation of quality
voice over a packet switched wireless network as governed by network quality metrics such as
latency, jitter and packet loss.
This research delivers a comprehensive analysis of 4th Generation (4G) networks such as,
Worldwide Interoperability for Microwave Access (WiMAX) and Long Term Evolution (LTE),
as specified by the standards bodies yet with focused orientation to the Quality of Service (QoS)
framework provided by each of the standards. Specific investigations are targeted towards the
Admission Control (AC) and Scheduling of physical resources over the air interface by the Media
Access Control (MAC) and Radio Link Control (RLC) layer. Current research and industry led
initiatives in the provisioning of quality voice, such as Circuit Switch Fallback (CSFB) and IP
Multimedia Subsystem (IMS) are presented and include the associated advantages and
disadvantages.
The results and recommendations of this research consist of a multi-faceted solution,
commencing with the addition of Heuristic Analysis with Deep Packet Inspection (DPI) being
proposed at the eNodeB or WiMAX Base Station (BS) level. An Admission Control (AC)
scheme tailored for voice utilising Heuristic Analysis as an input is created, thereafter an
identified QoS Class Identifier (QCI) Bearer or Service Flow and transportation Quality of
Service (QoS) Identifier for voice is triggered by the User Equipment (UE) application or Bearer
initiation procedures. The LTE Bearers and WiMAX Service Flows are tested with the intention
of recommending an LTE Bearer and WiMAX Service Flow that will ensure compliance to the
minimum required network quality metrics. Finally the testing of the invoking mechanisms is
presented mapping the Quality of Service (QoS) metrics across each of the network components
thereby completing the solution.
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Chiu, Chin-Lien, and 邱進連. "Joint call admission control/congestion control for wireless integrated voice/data networks." Thesis, 2001. http://ndltd.ncl.edu.tw/handle/53527107919468149937.
Full textAbstract:
碩士國立臺灣科技大學
電機工程系
89
The next generation wireless personal communication is expected to provide an efficient method that allows anybody to conduct multimedia communication in any place at any time. Different traffics usually have different transmission characteristics and quality of service requirements. Therefore, how to design an efficient media access control protocol is an important issue in terms of performance of wireless communication system. We consider wireless communication systems supporting both voice and data services, where call admission control (CAC) is used to guarantee that existing and new connections can have enough resource to maintain QoS at the connection level; and congestion control is used to guarantee QoS at the cell level. The call admission control adopted in the thesis is a simple threshold scheme, i.e., the number of voice terminals in use can not exceed the threshold. The congestion control used is D-TDMA/PRPTS, where data request packets can be transmitted in a minislot only if no voice request packets have already started at the same minislot. Therefore, collisions between voice request packets and data request packets can be avoided, the voice packet loss probability is reduced, and thus the capacity of voice terminals and channel utilization can be improved. Furthermore, to avoid excessive delay due to minislot contention, voice request packet is dropped if it can not succeed in the frame at which it originates. We study the effect of voice interarrival time and voice threshold on voice packet loss probability, voice call blocking probability, data packet average delay, and channel utilization. With a Markovian model, the voice packet loss probability, voice blocking probability and channel utilization of voice traffics are derived, and the analytical results are verified with the simulation results. Due to the mathematical complexity, for integrated voice/data scenarios, computer simulation is used to evaluate the effect of proposed CAC and congestion control on the performance measures of both voice and data traffics.
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Chen, Yao-Hsiang, and 陳耀翔. "Voice Remote Control Design of a Mobile Robot Using Shared-Control Approach." Thesis, 2017. http://ndltd.ncl.edu.tw/handle/7k3d3v.
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Sheng, Wu Wen, and 吳文生. "Identification and Control of Dual Voice Coil Speakers for Active Noise Control." Thesis, 2002. http://ndltd.ncl.edu.tw/handle/97107918216742236711.
Full textAbstract:
碩士國立中興大學
機械工程學系
90
This study aims to design a face velocity sensor of a dual voice coil speaker such that the sensor can later be used in feedback control of the speaker in active noise control applications to reduce associated pressure coupling effect. A structure of velocity sensor developed by Birdsong and Radcliffle [17] is adopted in this study to design the face velocity sensor that includes a current measurement, a voltage measurement and three sensor parameters. Analogy circuits for the current and voltage measurements are first designed and implemented. A frequency response method incorporating with a genetic algorithm is then developed to identify these sensor parameters such that a face velocity sensor can be obtained for experiments. This designed face velocity sensor is further implemented by using a digital signal processor. Proportion feedback controls based on the designed face velocity sensor of the dual voice coil connected to an acoustic duct are carried out, demonstrating the effectiveness of the designed face velocity sensor and the proposed identification technique in real applications.
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Tseng, Yu-Feng, and 曾渝楓. "Multisensor Based Intelligent HomMed Robot with Voice Control System." Thesis, 2007. http://ndltd.ncl.edu.tw/handle/02569659172288365781.
Full textAbstract:
碩士國立中正大學
電機工程所
95
With recently rapid growth of computer and robotic technology, the intelligent robot system has been applied to the industrial automation, hospital automation, museum guide, military application, home service, security service, and executing dangerous tasks for people. While the phenomenon which uses the intelligent robot is more extensive, it will promote the robotic research of international agitation, and contain the vast business opportunity. The main content of this thesis is to employ multi-sensor and voice system to apply in HomMed Robot (HMR). This thesis will illustrate these and show the experimental results both diversely and successfully. I devote to promote the voice system of HMR, and let HMR can execute various tasks more comfortable. The research objective of this thesis is to construct a multisensor based for intelligent HomMed robot with voice control. The voice control system includes the TTS (Text to Speech) system, voice commands recognition system, and robot control system. The TTS system transform texts include English and Chinese make our robot speak English and Chinese. The HomMed robot has the nature language to interact with people. Patients can use the voice control system to control the HomMed robot. The voice control system let the robot more convenient and comfortable. The experimental results for voice control system effectively improved the voice control ability of intelligent HomMed robot. Guarantee that intelligent HomMed robot can safely execute various kinds of task.
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Kao, Chien Chang, and 高健倡. "Apply Voice Recognition to Remote Control of Mechanical Horse." Thesis, 2004. http://ndltd.ncl.edu.tw/handle/99965409464953078254.
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Yang, Hsu, and 楊頊. "Model-Based Visual Feedback Control of Voice Coil Motors." Thesis, 2014. http://ndltd.ncl.edu.tw/handle/00228469811363733522.
Full textAbstract:
碩士國立臺灣海洋大學
電機工程學系
102
The implementation of visual feedback control systems is very difficult since the sampling rate is very low. In this thesis we consider the design of a visual feedback position controller for a voice coil motor (VCM) with a flexible load. By the distance information obtained from the visual sensor, a new model-based output feedback controller is designed for controlling the VCM to quickly move to a specified position. We use the optimal control theory, the regional pole placement approach, and the sliding mode control method, respectively, in the design of model-based controllers. We first calculate the controller parameters by Matlab software, and then implement the controller in a DSP28335 digital signal processor. Compared with the traditional state feedback controllers, the proposed model-based visual feedback controller provides a similar control performance.
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Wu, Chun-Feng, and 吳俊鋒. "QoS control for multi-stream voice over mobile IP networks." Thesis, 2011. http://ndltd.ncl.edu.tw/handle/93549915610519589730.
Full textAbstract:
博士國立交通大學
電信工程研究所
99
Packet loss and network delay are two essential problems to real-time voice communication over mobile IP Networks. The purpose of this dissertation is to develop a multi-stream voice communication system with its quality of service (QoS) control for increased channel robustness. The first part will focus on the error concealment of packet-erasure as well as channel bit errors. The basic strategy is a multiple description scalar quantization (MDSQ) system, in which multiple correlated indexes of the source are assigned and transmitted over channels to take advantage of largely uncorrelated loss and delay characteristics. We propose the use of turbo principle to develop a symbol-based iterative source-channel decoding algorithm for better decoding of multiple descriptions over a noisy channel. We first modify the BCJR algorithm based on sectionalization trellis so that symbol a posteriori probabilities can be derived and used as the extrinsic information to improve the iterative decoding between the source and channel decoders. The residual source redundancies are exploited as a priori informa- tion and a joint source decoding is formulated in the form of a maximum a posteriori estimation problem. We also formulate a recursive implementation for the source de- coder that processes reliability information received on different channels and combines them with inter-description correlation to estimate the transmitted quantizer indexes. Another important issue to address is the playout buffer design which is used at the receiver to smooth out the jitter. As a further step toward perceptual optimization, the error concealing capabilities of multiple description coding can be improved by including an forward error control (FEC) mechanism. We present an objective method for multi-stream voice quality prediction model. Based on the new prediction model, we proposed the use of minimum overall impairment as a perceptually motivated op- timization criterion for joint playout buffer and FEC control. Joint playout and FEC adjustment is then formulated as an optimization problem leading to a better balance between end-to-end delay and packet loss.
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Su, Meng-Kai, and 蘇盟凱. "An Electric Wheelchair System Based on Myoelectric and Voice Control." Thesis, 2000. http://ndltd.ncl.edu.tw/handle/87027063208591203095.
Full textAbstract:
碩士國立臺灣大學
電機工程學研究所
88
Electrical wheelchairs are the convenient tool for transportation of the physically disabled. However, the users need to use a joystick to control most of commercial wheelchairs. Consequently, those suffer from poor hand function can not use them smoothly. The goal of this study is to develop an electric wheelchair with voice control and myoelectric control for the disabled who lost their hand function. A voice recognition module which is based on filter bank technique is used for the purpose of voice control in this system. In myoelectric control, myoelectric signals associated with different motions are amplified and filtered via a special designed circuit for myoelectric signal. A recognition algorithm developed by our laboratory is used to recognize the filtered signal as the control command of this system. The main frame of this system is modified from a commercial electric wheelchair. The TMS320C32 Digital Signal Processor (DSP) is used to be the kernel of the control system. The DSP based control system performs the recognition algorithm for motion classification and controls the rotating speed and direction of two propulsive motors which drive the two rear wheels. The trajectory of this wheelchair is controlled via a proportional-integral-differential (PID) control algorithm according to the user’s desires.