Academic literature on the topic 'Voice signal transformation'
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Journal articles on the topic "Voice signal transformation"
Savic, Michael, and Il-Hyun Nam. "Voice personality transformation." Digital Signal Processing 1, no. 2 (April 1991): 107–10. http://dx.doi.org/10.1016/1051-2004(91)90099-7.
Full textTandyo, Anny, Martono Martono, and Adi Widyatmoko. "SPEAKER IDENTIFICATION MENGGUNAKAN TRANSFORMASI WAVELET DISKRIT DAN JARINGAN SARAF TIRUAN BACK-PROPAGATION." CommIT (Communication and Information Technology) Journal 2, no. 1 (May 31, 2008): 1. http://dx.doi.org/10.21512/commit.v2i1.482.
Full textZhao, Chun Hua, Chun Yu Ning, and Xiao Qiang Ji. "Design of Voice Prompt Temperature Detection System." Applied Mechanics and Materials 644-650 (September 2014): 1270–73. http://dx.doi.org/10.4028/www.scientific.net/amm.644-650.1270.
Full textWeb, B. W., and Ding Lin. "Transformation-based reconstruction for real-time voice transmissions over the Internet." IEEE Transactions on Multimedia 1, no. 4 (1999): 342–51. http://dx.doi.org/10.1109/6046.807954.
Full textTan, Choon Beng, Mohd Hanafi Ahmad Hijazi, Frazier Kok, Mohd Saberi Mohamad, and Puteri Nor Ellyza Nohuddin. "Artificial speech detection using image-based features and random forest classifier." IAES International Journal of Artificial Intelligence (IJ-AI) 11, no. 1 (March 1, 2022): 161. http://dx.doi.org/10.11591/ijai.v11.i1.pp161-172.
Full textBeltman, Willem, Hector Cordourier, and Paulo Lopez Meyer. "Hearing protection and communication in high noise environments using vibration sensing and neural network voice transformation." INTER-NOISE and NOISE-CON Congress and Conference Proceedings 263, no. 1 (August 1, 2021): 5027–37. http://dx.doi.org/10.3397/in-2021-2925.
Full textFong, Simon, Kun Lan, and Raymond Wong. "Classifying Human Voices by Using Hybrid SFX Time-Series Preprocessing and Ensemble Feature Selection." BioMed Research International 2013 (2013): 1–27. http://dx.doi.org/10.1155/2013/720834.
Full textGuimarães, Paula. "Retrieving Fin-de-Siècle Women Poets: The Transformative Myths, Fragments and Voices of Webster, Blind and Levy." Comparative Critical Studies 14, no. 2-3 (October 2017): 225–49. http://dx.doi.org/10.3366/ccs.2017.0237.
Full textIbrahim, Abu Bakar, and Ahmad Zamzuri Mohamad Ali. "Design of Microwave LNA Based on Ladder Matching Networks for WiMAX Applications." International Journal of Electrical and Computer Engineering (IJECE) 6, no. 4 (August 1, 2016): 1717. http://dx.doi.org/10.11591/ijece.v6i4.9877.
Full textIbrahim, Abu Bakar, and Ahmad Zamzuri Mohamad Ali. "Design of Microwave LNA Based on Ladder Matching Networks for WiMAX Applications." International Journal of Electrical and Computer Engineering (IJECE) 6, no. 4 (August 1, 2016): 1717. http://dx.doi.org/10.11591/ijece.v6i4.pp1717-1724.
Full textDissertations / Theses on the topic "Voice signal transformation"
Ardaillon, Luc. "Synthesis and expressive transformation of singing voice." Thesis, Paris 6, 2017. http://www.theses.fr/2017PA066511/document.
Full textThis 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
Degottex, Gilles. "Glottal source and vocal-tract separation : estimation of glottal parameters, voice transformation and synthesis using a glottal model." Paris 6, 2010. http://www.theses.fr/2010PA066399.
Full textLoscos, Àlex. "Spectral processing of the singing voice." Doctoral thesis, Universitat Pompeu Fabra, 2007. http://hdl.handle.net/10803/7542.
Full textLa tesi presenta nous procediments i formulacions per a la descripció i transformació d'aquells atributs específicament vocals de la veu cantada. La tesis inclou, entre d'altres, algorismes per l'anàlisi i la generació de desordres vocals como ara rugositat, ronquera, o veu aspirada, detecció i modificació de la freqüència fonamental de la veu, detecció de nasalitat, conversió de veu cantada a melodia, detecció de cops de veu, mutació de veu cantada, i transformació de veu a instrument; exemplificant alguns d'aquests algorismes en aplicacions concretes.
Esta tesis doctoral versa sobre el procesado digital de la voz cantada, más concretamente, sobre el análisis, transformación y síntesis de este tipo de voz basándose e dominio espectral, con especial énfasis en aquellas técnicas relevantes para el desarrollo de aplicaciones musicales.
La tesis presenta nuevos procedimientos y formulaciones para la descripción y transformación de aquellos atributos específicamente vocales de la voz cantada. La tesis incluye, entre otros, algoritmos para el análisis y la generación de desórdenes vocales como rugosidad, ronquera, o voz aspirada, detección y modificación de la frecuencia fundamental de la voz, detección de nasalidad, conversión de voz cantada a melodía, detección de los golpes de voz, mutación de voz cantada, y transformación de voz a instrumento; ejemplificando algunos de éstos en aplicaciones concretas.
This dissertation is centered on the digital processing of the singing voice, more concretely on the analysis, transformation and synthesis of this type of voice in the spectral domain, with special emphasis on those techniques relevant for music applications.
The thesis presents new formulations and procedures for both describing and transforming those attributes of the singing voice that can be regarded as voice specific. The thesis includes, among others, algorithms for rough and growl analysis and transformation, breathiness estimation and emulation, pitch detection and modification, nasality identification, voice to melody conversion, voice beat onset detection, singing voice morphing, and voice to instrument transformation; being some of them exemplified with concrete applications.
Calzada, Defez Àngel. "Conveying expressivity and vocal effort transformation in synthetic speech with Harmonic plus Noise Models." Doctoral thesis, Universitat Ramon Llull, 2016. http://hdl.handle.net/10803/360587.
Full textEsta tesis se llevó a cabo en el Grup en Tecnologies Mèdia de la Escuela de Ingeniería y Arquitectura la Salle. El grupo lleva una larga trayectoria dentro del campo de la síntesis de voz y cuenta con su propio sistema de síntesis por concatenación de unidades (US-TTS). El sistema permite sintetizar múltiples estilos expresivos mediante el uso de corpus específicos para cada estilo expresivo. De este modo, para realizar una síntesis agresiva, el sistema usa el corpus de este estilo, y para un estilo sensual, usa otro corpus específico para ese estilo. La presente tesis aborda el problema con un enfoque distinto proponiendo cambios en el esquema del sistema con el fin de mejorar la flexibilidad para sintetizar múltiples estilos expresivos a partir de un único corpus de estilo de habla neutro. El planteamiento seguido en esta tesis esta basado en el uso de técnicas de procesamiento de señales (DSP) para llevar a cabo modificaciones del señal de voz para que este exprese el estilo de habla deseado. Para llevar acabo las modificaciones de la señal de voz se han usado los modelos harmónico más ruido (HNM) por su flexibilidad para efectuar modificaciones de señales. La cualidad de la voz (VoQ) juega un papel importante en diferentes estilos expresivos. Por ello se exploró la síntesis expresiva basada en modificaciones de parámetros de bajo nivel de la VoQ. Durante este estudio se detectaron diferentes problemas que dieron pié a los objetivos planteados en esta tesis, entre ellos el encontrar un único parámetro con fuerte influencia en la expresividad. El parámetro seleccionado fue el esfuerzo vocal (VE) por su importante papel a la hora de expresar diferentes emociones. Las primeras pruebas se realizaron con el fin de transferir el VE entre dos realizaciones con diferente grado de VE de la misma palabra usando una metodología basada en un proceso filtrado de pre-émfasis adaptativo con coeficientes de predicción lineales (APLP). Esta primera aproximación logró transferir el nivel de VE entre dos realizaciones de la misma palabra, sin embargo el proceso presentaba limitaciones para generar niveles de esfuerzo vocal intermedios. A fin de mejorar la flexibilidad y el control del sistema para expresar diferentes niveles de VE, se planteó un nuevo modelo de VE basado en polinomios lineales. Este modelo permitió transferir el VE entre dos palabras diferentes e incluso generar nuevos niveles no presentes en el corpus usado para la síntesis. Esta flexibilidad está alineada con el objetivo general de esta tesis de permitir a un sistema US-TTS expresar múltiples estilos de habla expresivos a partir de un único corpus de estilo neutro. Además, la metodología propuesta incorpora un parámetro que permite de forma sencilla controlar el nivel de VE expresado en la voz sintetizada. Esto abre la posibilidad de controlar fácilmente el proceso de síntesis tal y como se hizo en el proyecto CreaVeu usando interfaces simples e intuitivas, también realizado dentro del grupo GTM. Esta memoria concluye con una revisión del trabajo realizado en esta tesis y con una propuesta de modificación de un esquema de US-TTS para expresar diferentes niveles de VE a partir de un único corpus neutro.
This thesis was conducted in the Grup en Tecnologies M`edia (GTM) from Escola d’Enginyeria i Arquitectura la Salle. The group has a long trajectory in the speech synthesis field and has developed their own Unit-Selection Text-To-Speech (US-TTS) which is able to convey multiple expressive styles using multiple expressive corpora, one for each expressive style. Thus, in order to convey aggressive speech, the US-TTS uses an aggressive corpus, whereas for a sensual speech style, the system uses a sensual corpus. Unlike that approach, this dissertation aims to present a new schema for enhancing the flexibility of the US-TTS system for performing multiple expressive styles using a single neutral corpus. The approach followed in this dissertation is based on applying Digital Signal Processing (DSP) techniques for carrying out speech modifications in order to synthesize the desired expressive style. For conducting the speech modifications the Harmonics plus Noise Model (HNM) was chosen for its flexibility in conducting signal modifications. Voice Quality (VoQ) has been proven to play an important role in different expressive styles. Thus, low-level VoQ acoustic parameters were explored for conveying multiple emotions. This raised several problems setting new objectives for the rest of the thesis, among them finding a single parameter with strong impact on the expressive style conveyed. Vocal Effort (VE) was selected for conducting expressive speech style modifications due to its salient role in expressive speech. The first approach working with VE was based on transferring VE between two parallel utterances based on the Adaptive Pre-emphasis Linear Prediction (APLP) technique. This approach allowed transferring VE but the model presented certain restrictions regarding its flexibility for generating new intermediate VE levels. Aiming to improve the flexibility and control of the conveyed VE, a new approach using polynomial model for modelling VE was presented. This model not only allowed transferring VE levels between two different utterances, but also allowed to generate other VE levels than those present in the speech corpus. This is aligned with the general goal of this thesis, allowing US-TTS systems to convey multiple expressive styles with a single neutral corpus. Moreover, the proposed methodology introduces a parameter for controlling the degree of VE in the synthesized speech signal. This opens new possibilities for controlling the synthesis process such as the one in the CreaVeu project using a simple and intuitive graphical interfaces, also conducted in the GTM group. The dissertation concludes with a review of the conducted work and a proposal for schema modifications within a US-TTS system for introducing the VE modification blocks designed in this dissertation.
Тодорів, Андрій Дмитрович. "Система багатофакторної аутентифікації користувачів комп’ютерних систем." Master's thesis, КПІ ім. Ігоря Сікорського, 2020. https://ela.kpi.ua/handle/123456789/38366.
Full textTopic relevance The solution to the problem of corporate data protection in the XXI century has gone beyond the physical interaction with employees, due to the transition of the required information into a computer format. This feature has formed the need to develop and implement new mechanisms for corporate data protection. The proposed system of authentication of computer system users, developed on the basis of neural network technologies, provides the possibility of user identification on the basis of individual anthropometric visual and voice indicators of the subject, in order to prevent theft of corporate data and identification of criminal entities. The object of study is the transformation of anthropometric indicators into a computer form. The subject of study is the mechanisms of pattern recognition. The goal of this work is to improve the capabilities of biometric identification methods of subjects by developing a new architecture based on neural networks. Study methods. Comparison of existing algorithms on the criteria of accuracy, speed, resource costs, reliability, in order to implement and further modify the corporate control system. The scientific novelty is the development of a new mechanism for identifying subjects that combines algorithms for voice and visual identification of subjects. The practical value lies in the possibility of using this system in a corporate environment in order to prevent data leakage and identification of criminal entities. Low resource consumption contributes to the application of the developed algorithm in highly loaded systems. Structure and scope of work. The master's dissertation consists of an introduction, four chapters, conclusions and appendices. The introduction analyzes the problem of corporate data protection. The prospects of using the mechanisms of biometric voice and visual identification of subjects for its solution are substantiated. Biometric identification algorithms are investigated. The first section describes the existing algorithms for recognizing visual and voice images. The second section investigates the feasibility of using existing algorithms for voice and visual biometric identification, analyzes and compares existing image recognition architectures. The third section describes the process of developing algorithms for visual and voice biometric user identification The fourth section presents the characteristics of the developed COP, the test results, the system is studied on different data sets, and its modification in order to achieve the specified accuracy. The conclusions summarize the results of research and development.
Book chapters on the topic "Voice signal transformation"
Alsubari, Akram, Ghanshyam D. Ramteke, and Rakesh J. Ramteke. "Transformation of Voice Signals to Spatial Domain for Code Optimization in Digital Image Processing." In Communications in Computer and Information Science, 196–209. Singapore: Springer Singapore, 2021. http://dx.doi.org/10.1007/978-981-16-0493-5_18.
Full textKumekawa, Ian. "Retreat to the Ivory Tower." In The First Serious Optimist. Princeton University Press, 2017. http://dx.doi.org/10.23943/princeton/9780691163482.003.0006.
Full textGaines, Malik. "The Cockettes, Sylvester, and Performance as Life." In Black Performance on the Outskirts of the Left. NYU Press, 2017. http://dx.doi.org/10.18574/nyu/9781479837038.003.0005.
Full textConference papers on the topic "Voice signal transformation"
Stylianou, Yannis. "Voice Transformation: A survey." In ICASSP 2009 - 2009 IEEE International Conference on Acoustics, Speech and Signal Processing. IEEE, 2009. http://dx.doi.org/10.1109/icassp.2009.4960401.
Full textJin, Qin, Arthur R. Toth, Tanja Schultz, and Alan W. Black. "Voice convergin: Speaker de-identification by voice transformation." In ICASSP 2009 - 2009 IEEE International Conference on Acoustics, Speech and Signal Processing. IEEE, 2009. http://dx.doi.org/10.1109/icassp.2009.4960482.
Full textValbret, H., E. Moulines, and J. P. Tubach. "Voice transformation using PSOLA technique." In [Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing. IEEE, 1992. http://dx.doi.org/10.1109/icassp.1992.225951.
Full textPopa, Victor, Hanna Silen, Jani Nurminen, and Moncef Gabbouj. "Local linear transformation for voice conversion." In ICASSP 2012 - 2012 IEEE International Conference on Acoustics, Speech and Signal Processing. IEEE, 2012. http://dx.doi.org/10.1109/icassp.2012.6288922.
Full textSisman, Berrak, Haizhou Li, and Kay Chen Tan. "Transformation of prosody in voice conversion." In 2017 Asia-Pacific Signal and Information Processing Association Annual Summit and Conference (APSIPA ASC). IEEE, 2017. http://dx.doi.org/10.1109/apsipa.2017.8282288.
Full textFernandez, Raul, Andrew Rosenberg, Alexander Sorin, Bhuvana Ramabhadran, and Ron Hoory. "Voice-transformation-based data augmentation for prosodic classification." In 2017 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP). IEEE, 2017. http://dx.doi.org/10.1109/icassp.2017.7953214.
Full textTakahashi, Masahito, and Hiroki Matsumoto. "A consideration on the method of transformation from whisper voice to voice sounds." In 2011 International Symposium on Intelligent Signal Processing and Communications Systems (ISPACS 2011). IEEE, 2011. http://dx.doi.org/10.1109/ispacs.2011.6146160.
Full textQin Jin, Arthur R. Toth, Alan W. Black, and Tanja Schultz. "Is voice transformation a threat to speaker identification?" In ICASSP 2008 - 2008 IEEE International Conference on Acoustics, Speech and Signal Processing. IEEE, 2008. http://dx.doi.org/10.1109/icassp.2008.4518742.
Full textTurk, Oytun, Osman Buyuk, Ali Haznedaroglu, and Levent M. Arslan. "Application of voice conversion for cross-language rap singing transformation." In ICASSP 2009 - 2009 IEEE International Conference on Acoustics, Speech and Signal Processing. IEEE, 2009. http://dx.doi.org/10.1109/icassp.2009.4960404.
Full textChithra, PL, and R. Aparna. "Voice Signal Encryption Scheme Using Transformation and Embedding Techniques for Enhanced Security." In 2018 2nd International Conference on Imaging, Signal Processing and Communication (ICISPC). IEEE, 2018. http://dx.doi.org/10.1109/icispc44900.2018.9006681.
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