Relevant bibliographies by topics / Vocoders

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  1. Journal articles
  2. Dissertations / Theses
  3. Books
  4. Book chapters
  5. Conference papers
  6. Reports

Academic literature on the topic 'Vocoders'

Author: Grafiati
Published: 4 June 2021
Last updated: 30 January 2023

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Journal articles on the topic "Vocoders"

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Karoui, Chadlia, Chris James, Pascal Barone, David Bakhos, Mathieu Marx, and Olivier Macherey. "Searching for the Sound of a Cochlear Implant: Evaluation of Different Vocoder Parameters by Cochlear Implant Users With Single-Sided Deafness." Trends in Hearing 23 (January 2019): 233121651986602. http://dx.doi.org/10.1177/2331216519866029.

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Cochlear implantation in subjects with single-sided deafness (SSD) offers a unique opportunity to directly compare the percepts evoked by a cochlear implant (CI) with those evoked acoustically. Here, nine SSD-CI users performed a forced-choice task evaluating the similarity of speech processed by their CI with speech processed by several vocoders presented to their healthy ear. In each trial, subjects heard two intervals: their CI followed by a certain vocoder in Interval 1 and their CI followed by a different vocoder in Interval 2. The vocoders differed either (i) in carrier type—(sinusoidal [SINE], bandfiltered noise [NOISE], and pulse-spreading harmonic complex) or (ii) in frequency mismatch between the analysis and synthesis frequency ranges—(no mismatch, and two frequency-mismatched conditions of 2 and 4 equivalent rectangular bandwidths [ERBs]). Subjects had to state in which of the two intervals the CI and vocoder sounds were more similar. Despite a large intersubject variability, the PSHC vocoder was judged significantly more similar to the CI than SINE or NOISE vocoders. Furthermore, the No-mismatch and 2-ERB mismatch vocoders were judged significantly more similar to the CI than the 4-ERB mismatch vocoder. The mismatch data were also interpreted by comparing spiral ganglion characteristic frequencies with electrode contact positions determined from postoperative computed tomography scans. Only one subject demonstrated a pattern of preference consistent with adaptation to the CI sound processor frequency-to-electrode allocation table and two subjects showed possible partial adaptation. Those subjects with adaptation patterns presented overall small and consistent frequency mismatches across their electrode arrays.
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Roebel, Axel, and Frederik Bous. "Neural Vocoding for Singing and Speaking Voices with the Multi-Band Excited WaveNet." Information 13, no. 3 (2022): 103. http://dx.doi.org/10.3390/info13030103.

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The use of the mel spectrogram as a signal parameterization for voice generation is quite recent and linked to the development of neural vocoders. These are deep neural networks that allow reconstructing high-quality speech from a given mel spectrogram. While initially developed for speech synthesis, now neural vocoders have also been studied in the context of voice attribute manipulation, opening new means for voice processing in audio production. However, to be able to apply neural vocoders in real-world applications, two problems need to be addressed: (1) To support use in professional audio workstations, the computational complexity should be small, (2) the vocoder needs to support a large variety of speakers, differences in voice qualities, and a wide range of intensities potentially encountered during audio production. In this context, the present study will provide a detailed description of the Multi-band Excited WaveNet, a fully convolutional neural vocoder built around signal processing blocks. It will evaluate the performance of the vocoder when trained on a variety of multi-speaker and multi-singer databases, including an experimental evaluation of the neural vocoder trained on speech and singing voices. Addressing the problem of intensity variation, the study will introduce a new adaptive signal normalization scheme that allows for robust compensation for dynamic and static gain variations. Evaluations are performed using objective measures and a number of perceptual tests including different neural vocoder algorithms known from the literature. The results confirm that the proposed vocoder compares favorably to the state-of-the-art in its capacity to generalize to unseen voices and voice qualities. The remaining challenges will be discussed.
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Harding, Eleanor E., Etienne Gaudrain, Imke J. Hrycyk, et al. "Musical Emotion Categorization with Vocoders of Varying Temporal and Spectral Content." Trends in Hearing 27 (January 2023): 233121652211411. http://dx.doi.org/10.1177/23312165221141142.

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While previous research investigating music emotion perception of cochlear implant (CI) users observed that temporal cues informing tempo largely convey emotional arousal (relaxing/stimulating), it remains unclear how other properties of the temporal content may contribute to the transmission of arousal features. Moreover, while detailed spectral information related to pitch and harmony in music — often not well perceived by CI users— reportedly conveys emotional valence (positive, negative), it remains unclear how the quality of spectral content contributes to valence perception. Therefore, the current study used vocoders to vary temporal and spectral content of music and tested music emotion categorization (joy, fear, serenity, sadness) in 23 normal-hearing participants. Vocoders were varied with two carriers (sinewave or noise; primarily modulating temporal information), and two filter orders (low or high; primarily modulating spectral information). Results indicated that emotion categorization was above-chance in vocoded excerpts but poorer than in a non-vocoded control condition. Among vocoded conditions, better temporal content (sinewave carriers) improved emotion categorization with a large effect while better spectral content (high filter order) improved it with a small effect. Arousal features were comparably transmitted in non-vocoded and vocoded conditions, indicating that lower temporal content successfully conveyed emotional arousal. Valence feature transmission steeply declined in vocoded conditions, revealing that valence perception was difficult for both lower and higher spectral content. The reliance on arousal information for emotion categorization of vocoded music suggests that efforts to refine temporal cues in the CI user signal may immediately benefit their music emotion perception.
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Bernstein, Lynne E., Marilyn E. Demorest, Michael P. O'Connell, and David C. Coulter. "Lipreading with vibrotactile vocoders." Journal of the Acoustical Society of America 87, S1 (1990): S124—S125. http://dx.doi.org/10.1121/1.2027907.

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Mohammed, Zinah J., and Abdulkareem A. Kadhim. "A Comparative Study of Speech Coding Techniques for Electro Larynx Speech Production." Iraqi Journal of Information and Communication Technology 5, no. 1 (2022): 31–41. http://dx.doi.org/10.31987/ijict.5.1.185.

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Speech coding is a method of earning a tight speech signals representation for efficient storage and efficient transmission over band limited wired or wireless channels. This is usually achieved with acceptable representation and least number of bits without depletion in the perceptual quality. A number of speech coding methods already developed and various speech coding algorithms for speech analysis and synthesis are used. This paper deals with the comparison of selected coding methods for speech signal produced by Electro Larynx (EL) device. The latter is a device used by cancer patients with their vocal laryngeal cords being removed. The used methods are Residual-Excited Linear Prediction (RELP), Code Excited Linear Prediction (CELP), Algebraic Code Excited Linear Predictive (ACELP), Phase Vocoders based on Wavelet Transform (PVWT), Channel Vocoders based on Wavelet Transform (CVWT), and Phase vocoder based on Dual-Tree Rational-Dilation Complex Wavelet Transform (PVDT-RADWT). The aim here is to select the best coding approach based on the quality of the reproduced speech. The signal used in the test is speech signal recorded either directly by normal persons or else produced by EL device. The performance of each method is evaluated using both objective and subjective listening tests. The results indicate that PVWT and ACELP coders perform better than other methods having about 40 dB SNR and 3 PESQ score for EL speech and 75 dB with 3.5 PESQ score for normal speech, respectively.
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Ausili, Sebastian A., Bradford Backus, Martijn J. H. Agterberg, A. John van Opstal, and Marc M. van Wanrooij. "Sound Localization in Real-Time Vocoded Cochlear-Implant Simulations With Normal-Hearing Listeners." Trends in Hearing 23 (January 2019): 233121651984733. http://dx.doi.org/10.1177/2331216519847332.

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Bilateral cochlear-implant (CI) users and single-sided deaf listeners with a CI are less effective at localizing sounds than normal-hearing (NH) listeners. This performance gap is due to the degradation of binaural and monaural sound localization cues, caused by a combination of device-related and patient-related issues. In this study, we targeted the device-related issues by measuring sound localization performance of 11 NH listeners, listening to free-field stimuli processed by a real-time CI vocoder. The use of a real-time vocoder is a new approach, which enables testing in a free-field environment. For the NH listening condition, all listeners accurately and precisely localized sounds according to a linear stimulus–response relationship with an optimal gain and a minimal bias both in the azimuth and in the elevation directions. In contrast, when listening with bilateral real-time vocoders, listeners tended to orient either to the left or to the right in azimuth and were unable to determine sound source elevation. When listening with an NH ear and a unilateral vocoder, localization was impoverished on the vocoder side but improved toward the NH side. Localization performance was also reflected by systematic variations in reaction times across listening conditions. We conclude that perturbation of interaural temporal cues, reduction of interaural level cues, and removal of spectral pinna cues by the vocoder impairs sound localization. Listeners seem to ignore cues that were made unreliable by the vocoder, leading to acute reweighting of available localization cues. We discuss how current CI processors prevent CI users from localizing sounds in everyday environments.
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Ozdamar, Ozcan, Rebecca E. Eilers, and D. Kimbrough Oller. "Tactile Vocoders for the Deaf." IEEE Engineering in Medicine and Biology Magazine 6, no. 3 (1987): 37–42. http://dx.doi.org/10.1109/memb.1987.5006436.

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VAN RENSBURG, T. JANSE, M. A. VAN WYK, A. T. POTGIETER, and W. H. STEEB. "PHASE VOCODER TECHNOLOGY FOR THE SIMULATION OF ENGINE SOUND." International Journal of Modern Physics C 17, no. 05 (2006): 721–31. http://dx.doi.org/10.1142/s0129183106009333.

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For a driving simulator which should be an exact replica of a certain vehicle, an accurate sound model is of extreme importance. The most games select between three or more prerecorded engine sounds, depending on the engine speed. Other methods use linear interpolation between engine sounds for a more accurate approximation, but this is still not ideal. By using vocoders, a technique used for the manipulation of voice, a much higher level of accuracy and realism can be obtained. This article proposes the use of vocoders for the modeling of engine sound for driving simulation and computer driving games.
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Fodor, Ádám, László Kopácsi, Zoltán Ádám Milacski, and András Lőrincz. "Speech De-identification with Deep Neural Networks." Acta Cybernetica 25, no. 2 (2021): 257–69. http://dx.doi.org/10.14232/actacyb.288282.

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Cloud-based speech services are powerful practical tools but the privacy of the speakers raises important legal concerns when exposed to the Internet. We propose a deep neural network solution that removes personal characteristics from human speech by converting it to the voice of a Text-to-Speech (TTS) system before sending the utterance to the cloud. The network learns to transcode sequences of vocoder parameters, delta and delta-delta features of human speech to those of the TTS engine. We evaluated several TTS systems, vocoders and audio alignment techniques. We measured the performance of our method by (i) comparing the result of speech recognition on the de-identified utterances with the original texts, (ii) computing the Mel-Cepstral Distortion of the aligned TTS and the transcoded sequences, and (iii) questioning human participants in A-not-B, 2AFC and 6AFC tasks. Our approach achieves the level required by diverse applications.
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Chang, W. W., and D. Y. Wang. "Quality enhancement of sinusoidal transform vocoders." IEE Proceedings - Vision, Image, and Signal Processing 145, no. 6 (1998): 379. http://dx.doi.org/10.1049/ip-vis:19982456.

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Dissertations / Theses on the topic "Vocoders"

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Crossman, A. H. "Multipulse-excitation applied to vocoders." Thesis, University of Cambridge, 1987. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.232981.

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Multipulse-excitation has greatly improved the speech quality achievable from linear predictive coders which previously required speech to be classified as voiced or unvoiced for excitation purposes. Multipulse removes the need for voicing classification, improving speech quality by enhancing the excitation and offsetting errors in the vocal tract filter. An investigation of multipulse-excitation applied to a channel vocoder and a formant synthesiser was conducted. The prime objective was to improve the performance of these algorithms and achieve multipulse linear prediction speech quality, our target quality. This dissertation outlines and restates the idea of multipulse-excitation applied to a linear predictive vocoder. We then examine a high quality channel vocoder and formant synthesiser, and the use of multipulse-excitation to improve their performances. In each case time and frequency domain multipulsecalgorithms were used. Various modifications were made to these algorithms in order to accommodate multipulse-excitation and improve the overall speech quality. In the case of the channel vocoder this involved a novel technique, which sacrificed the inherent waveform preserving properties of the multipulse algorithm. Only by increasing both the pulse rate and the number of channels could the multipulse-excited channel vocoder achieve our target quality. With the formant synthesiser it was possible, by variation of the pulse rate alone, to achieve our target quality. Comparisons are drawn between the three multipulse algorithms and reasons given for their differing performance; this is substantiated by experimental results. These results suggested interesting improvements to the multipulse-excited formant synthesiser; and also hinted at a new and novel technique for formant tracking, using multipulse-excitation applied to a formant synthesiser.
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Ma, Wei. "Multi-band excitation based vocoders and their real-time implementation." Thesis, University of Surrey, 1994. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.240182.

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Fang, Jie. "Design of secure speech encryption systems." Thesis, Queensland University of Technology, 1990. https://eprints.qut.edu.au/36471/1/36471_Fang_1990.pdf.

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This thesis investigates the design of digital speech encryption systems based on low bit rate vocoders. The speech quality and the cryptographic strength of the system are determined by vocoder and encryptor respectively. Three different low bit rate vocoders, 2400 BPS LPC ( Linear Prediction Coding) vocoder, 9600 BPS MELPC (Mul tipulse Excited Linear Prediction Coding) vocoder and 4800 BPS CELP (Codebook Excited Linear Prediction coding) vocoder, have been simulated. The performances of these vocoders are evaluated by using four objective measures. The thesis considers the follows aspects of digital encryption system: * Security * Speech quality * Robustness * System delay Several choices of the cryptosystem for the encryption of digital speech are investigated, and the performance of the overall system is discussed. The work presented in this thesis enables a secure communication system designer to select a speech coding scheme and a cipher system to meet required level of security and speech quality. encryption systems throughout this thesis refers to mathematics analysis and simulation of such systems rather than the actual construction of electronic circuits.
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Bliūdžius, Mindaugas. "Skaitmeninių kalbos įrašų glaudinimo metodai." Master's thesis, Lithuanian Academic Libraries Network (LABT), 2004. http://vddb.library.lt/obj/LT-eLABa-0001:E.02~2004~D_20040529_122424-17577.

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The past three decades has witnessed substantial progress towards the application of low-rate speech coders to civilian and military communications as well as computer-related voice applications. Central to this progress has been the development of new speech coders capable of producing high-quality speech at low data rates. Most of these coders incorporate mechanisms to: represent the spectral properties of speech, provide for speech waveform matching, and "optimize" the coder's performance for the human ear. A number of these coders have already been adopted in national and international cellular telephony standards. The objective of this paper is to provide a tutorial overview of speech coding methodologies with emphasis on those algorithms that are part of the recent low-rate standards for voice applications. Although the emphasis is on the new low-rate coders, we attempt to provide a comprehensive survey by covering some of the traditional methodologies as well. The paper starts with a historical perspective and continues with a brief discussion on the speech properties and performance measures. Then I proceed with descriptions of waveform coders, linear predictive vocoders, and analysis-by-synthesis linear predictive coders. At the end the system for computer-based stenographing is presented. Quality research and ways how to improve this system will be provided.
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Chmayssani, Toufic. "Modulation sur les canaux vocodés." Phd thesis, Université Paris-Est, 2010. http://tel.archives-ouvertes.fr/tel-00587629.

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Les canaux vocodés sont les canaux de communications dédiés à la voix et dans lesquels le signal traverse divers équipements destinés au transport de la voix tels que des codeurs de parole, des détecteurs d'activité vocale (VAD), des systèmes de transmission discontinue (DTX). Il peut s'agir de systèmes de communications téléphoniques filaires ou mobiles (réseaux cellulaires 2G/3G, satellites INMARSAT...) ou de voix sur IP. Les codeurs de parole dans les normes récentes pour les réseaux de téléphonie mobiles ou de voix sur IP font appel à des algorithmes de compression dérivés de la technique CELP (Code Excited Linear Prediction) qui permettent d'atteindre des débits de l'ordre de la dizaine de Kb/s bien inférieurs aux codeurs des réseaux téléphoniques filaires (typiquement 64 ou 32 Kb/s). Ces codeurs tirent leur efficacité de l'utilisation de caractéristiques spécifiques aux signaux de parole et à l'audition humaine. Aussi les signaux autres que la parole sont-ils généralement fortement distordus par ces codeurs. La transmission de données sur les canaux vocodés peut être intéressante pour des raisons liées à la grande disponibilité des canaux dédiés à la voix et pour des raisons de discrétion de la communication (sécurité). Mais le signal modulé transmis sur ces canaux vocodés est soumis aux dégradations causées par les codeurs de parole, ce qui impose des contraintes sur le type de modulation utilisé. Cette thèse a porté sur la conception et l'évaluation de modulations permettant la transmission de données sur les canaux vocodés. Deux approches de modulations ont été proposées pour des applications correspondant à des débits de transmission possibles assez différents. La principale application visée par la thèse concerne la transmission de parole chiffrée, transmission pour laquelle le signal de parole est numérisé, comprimé à bas débit par un codeur de parole puis sécurisé par un algorithme de cryptage. Pour cette application, nous nous sommes focalisés sur les réseaux de communications utilisant des codeurs CELP de débits supérieurs à la dizaine de Kb/s typiquement les canaux de communication mobiles de deuxième ou troisième génération. La première approche de modulation proposée concerne cette application. Elle consiste à utiliser des modulations numériques après optimisation de leurs paramètres de façon à prendre en compte les contraintes imposées par le canal et à permettre des débits et des performances en probabilité d'erreur compatibles avec la transmission de parole chiffrée (typiquement un débit supérieur à 1200 b/s avec un BER de l'ordre de 10-3). Nous avons montré que la modulation QPSK optimisée permet d'atteindre ces performances. Un système de synchronisation est aussi étudié et adapté aux besoins et aux contraintes du canal vocodé. Les performances atteintes par la modulation QPSK avec le système de synchronisation proposé, ainsi que la qualité de la parole sécurisée transmise ont été évalués par simulation et validés expérimentalement sur un canal GSM réel grâce à un banc de test développé dans la thèse.La deuxième approche de modulation a privilégié la robustesse du signal modulé lors de la transmission à travers un codeur de parole quelconque, même un codeur à bas débit tels que les codeurs MELP à 2400 ou 1200 b/s. Dans ce but, nous avons proposé une modulation effectuée par concaténation de segments de parole naturelle associée à une technique de démodulation qui segmente le signal reçu et identifie les segments de parole par programmation dynamique avec taux de reconnaissance élevé. Cette modulation a été évaluée par simulation sur différents codeurs de parole. Elle a aussi été testée sur des canaux GSM réels. Les résultats obtenus montrent une probabilité d'erreur très faible quelque soit le canal vocodé et le débit des codeurs de parole utilisés mais pour des débits possibles relativement faibles. Les applications envisageables sont restreintes à des débits typiquement inférieurs à 200 b/s.Enfin nous nous sommes intéressés aux détecteurs d'activité vocale dont l'effet peut-être très dommageable pour les signaux de données. Nous avons proposé une méthode permettant de contrer les VAD utilisés dans les réseaux GSM. Son principe consiste à rompre la stationnarité du spectre du signal modulé, stationnarité sur laquelle s'appuie le VAD pour décider que le signal n'est pas de la parole
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LeBlanc, Wilfrid P. (Wilfrid Paul) Carleton University Dissertation Engineering Electrical. "An advanced speech coder based on a rate-distortion theory framework." Ottawa, 1988.

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Griffin, Daniel W. "Multi-band excitation vocoder." Thesis, Massachusetts Institute of Technology, 1987. http://hdl.handle.net/1721.1/14803.

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Martins, José Antônio. "Vocoder LPC com quantização vetorial." [s.n.], 1991. http://repositorio.unicamp.br/jspui/handle/REPOSIP/261389.

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Orientador : Fabio Violaro<br>Dissertação (mestrado) - Universidade Estadual de Campinas, Faculdade de Engenharia Eletrica<br>Made available in DSpace on 2018-07-13T23:59:40Z (GMT). No. of bitstreams: 1 Martins_JoseAntonio_M.pdf: 6784204 bytes, checksum: 4e9df50ca8f72e1710d541924b76a67c (MD5) Previous issue date: 1991<br>Resumo: Neste trabalho são descritos os princípios do vocoder LPC, sendo mostrados os métodos para cálculo dos parâmetros do mesmo. Também são apresentados os resultados de simulações de vocoders LPC usando quantização escalar, quantização vetorial e interpolação dos parâmetros quantizados. Inicialmente foi projetado um vocoder LPC não quantizado, o qual serviu de padrão para a avaliação dos vocoders quantizados. Usando a quantização escalar dos coeficientes razão log-área foi obtido um vocoder à taxa de 2200 bit /s, assegurando uma boa qualidade e alta inteligibilidade da voz sintetizada. Com o uso da quantização vetorial obteve-se um bom desempenho em taxas da ordem de 1000 bit/s. Essas taxas foram reduzidas em 50% com o uso da interpolação linear, transmitindo apenas os parâmetros dos quadros ímpares. Assim, conseguiu-se vocoders com taxas ao redor de 500 bit/s, apresentando voz sintetizada com degradação em relação aos sistemas anteriores, mas ainda assegurando uma boa inteligibilidade<br>Abstract: Not informed.<br>Mestrado<br>Eletronica e Comunicações<br>Mestre em Engenharia Elétrica
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Vávra, Jakub. "Šifrování telefonních hovorů." Master's thesis, Vysoké učení technické v Brně. Fakulta informačních technologií, 2008. http://www.nusl.cz/ntk/nusl-235954.

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This master's thesis is about making draft and implementing land-line phone call encryption using FITkit. The ultimate goal is to find suitable compression and encryption methods, implement or adapt them for FITkit board and create functional solution.
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Hudson, Nicholaus D. W. "The self-excited vocoder for mobile telephony." Thesis, University of Bath, 1992. https://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.760629.

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Books on the topic "Vocoders"

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Redding, Christopher. Voice quality assessment of vocoders in tandem configuration. U.S. Dept. of Commerce, National Telecommunications and Information Administration, 2001.

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Surphlis, David S. Multi-band excitation vocoder. The Author], 1996.

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Moore, James Thomas. A mixed excitation vocoder with fuzzy logic classifier. Naval Postgraduate School, 1992.

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Papamichalis, Panos E. Practical approaches to speech coding. Prentice-Hall, 1987.

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Tompkins, Dave. How to wreck a nice beach: The vocoder from World War II to hip-hop : the machine speaks. Melville House, 2010.

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Tompkins, Dave. How to wreck a nice beach: The vocoder from World War II to hip-hop : the machine speaks. Melville House, 2011.

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Fundamentals of voice-quality engineering in wireless networks. Cambridge University Press, 2007.

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How to wreck a nice beach: The vocoder from World War II to hip-hop : the machine speaks. Stop Smiling/Melville House Pub., 2010.

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Ramamurthy, Karthikeyan N. MATLAB software for the code excited linear prediction algorithm: The Federal Standard, 1016. Morgan & Claypool Publishers, 2010.

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Voice quality assessment of vocoders in tandem configuration. U.S. Dept. of Commerce, National Telecommunications and Information Administration, 2001.

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Book chapters on the topic "Vocoders"

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Rao, R. Chinna, D. Elizabath Rani, and S. Srinivasa Rao. "Basic Framework of Vocoders for Speech Processing." In Advances in Intelligent Systems and Computing. Springer Singapore, 2019. http://dx.doi.org/10.1007/978-981-13-3393-4_66.

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Yoshida, Satoshi, Ken’ichi Furuya, and Hideyuki Mizuno. "Introducing Speaker Vectors for Child Speech Synthesis in Neural Vocoders." In Complex, Intelligent and Software Intensive Systems. Springer International Publishing, 2022. http://dx.doi.org/10.1007/978-3-031-08812-4_52.

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Bhatia, Rhythm Rajiv, and Tomi H. Kinnunen. "An Initial Study on Birdsong Re-synthesis Using Neural Vocoders." In Speech and Computer. Springer International Publishing, 2022. http://dx.doi.org/10.1007/978-3-031-20980-2_7.

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Weik, Martin H. "vocoder." In Computer Science and Communications Dictionary. Springer US, 2000. http://dx.doi.org/10.1007/1-4020-0613-6_20884.

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Sheidaei, Shervin, Hamid Noori, Ahmad Akbari, and Hosein Pedram. "Motivation from a Full-Rate Specific Design to a DSP Core Approach for GSM Vocoders." In Field-Programmable Logic and Applications. Springer Berlin Heidelberg, 2001. http://dx.doi.org/10.1007/3-540-44687-7_40.

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Yoshida, Satoshi, Shingo Uenohara, Keisuke Nishijima, and Ken’ichi Furuya. "Voice Quality Change Due to the Amount of Training Data for Multi- and Target-Speaker WaveNet Vocoders." In Complex, Intelligent and Software Intensive Systems. Springer International Publishing, 2021. http://dx.doi.org/10.1007/978-3-030-79725-6_73.

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Chung, Jae H., and Ronald W. Schafer. "Vector Excitation Homomorphic Vocoder." In Advances in Speech Coding. Springer US, 1991. http://dx.doi.org/10.1007/978-1-4615-3266-8_23.

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Gerstlauer, Andreas. "Design of a GSM Vocoder." In System Design. Springer US, 2001. http://dx.doi.org/10.1007/978-1-4615-1481-7_3.

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Pirkle, Will C. "FFT Processing: The Phase Vocoder." In Designing Audio Effect Plugins in C++. Routledge, 2019. http://dx.doi.org/10.4324/9780429490248-20.

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Vít, Jakub, Zdeněk Hanzlíček, and Jindřich Matoušek. "Czech Speech Synthesis with Generative Neural Vocoder." In Text, Speech, and Dialogue. Springer International Publishing, 2019. http://dx.doi.org/10.1007/978-3-030-27947-9_26.

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Conference papers on the topic "Vocoders"

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Maiti, Soumi, and Michael I. Mandel. "Parametric Resynthesis With Neural Vocoders." In 2019 IEEE Workshop on Applications of Signal Processing to Audio and Acoustics (WASPAA). IEEE, 2019. http://dx.doi.org/10.1109/waspaa.2019.8937165.

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Ribeiro, Carlos M., and Isabel M. Trancoso. "Improving speaker recognisability in phonetic vocoders." In 5th International Conference on Spoken Language Processing (ICSLP 1998). ISCA, 1998. http://dx.doi.org/10.21437/icslp.1998-396.

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Hoffmann, Rudiger. "On the development of early vocoders." In 2010 Second IEEE Region 8 Conference on the History of Telecommunications (HISTELCON). IEEE, 2010. http://dx.doi.org/10.1109/histelcon.2010.5735282.

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Onishi, Kotaro, and Toru Nakashika. "Consistency Regularization for GAN-based Neural Vocoders." In 2022 Asia Pacific Signal and Information Processing Association Annual Summit and Conference (APSIPA ASC). IEEE, 2022. http://dx.doi.org/10.23919/apsipaasc55919.2022.9980309.

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Chung, J. H., and R. W. Schafer. "Performance evaluation of analysis-by-synthesis homomorphic vocoders." In [Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing. IEEE, 1992. http://dx.doi.org/10.1109/icassp.1992.226106.

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Mustafa Badi, Zaha, and Lamia Fathi Abusedra. "Neural Network-based Vocoders in Arabic Speech Synthesis." In ICEMIS'21: The 7th International Conference on Engineering & MIS 2021. ACM, 2021. http://dx.doi.org/10.1145/3492547.3492623.

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Florencio, D. A. F. "Investigating the use of asymmetric windows in CELP vocoders." In Proceedings of ICASSP '93. IEEE, 1993. http://dx.doi.org/10.1109/icassp.1993.319331.

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Song, Eunwoo, Jin-Seob Kim, Kyungguen Byun, and Hong-Goo Kang. "Speaker-Adaptive Neural Vocoders for Parametric Speech Synthesis Systems." In 2020 IEEE 22nd International Workshop on Multimedia Signal Processing (MMSP). IEEE, 2020. http://dx.doi.org/10.1109/mmsp48831.2020.9287168.

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Cavalcante, Dirceu, and Rafael Lins. "A Comparative Analysis of Vocoders for Communications in Portuguese." In VII International Telecommunications Symposium. Sociedade Brasileira de Telecomunicações, 2010. http://dx.doi.org/10.14209/sbrt.2010.65.

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Okamoto, Takuma, Tomoki Toda, Yoshinori Shiga, and Hisashi Kawai. "Noise Level Limited Sub-Modeling for Diffusion Probabilistic Vocoders." In ICASSP 2021 - 2021 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP). IEEE, 2021. http://dx.doi.org/10.1109/icassp39728.2021.9415087.

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Reports on the topic "Vocoders"

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Li, A. RTP Payload Format for Enhanced Variable Rate Codecs (EVRC) and Selectable Mode Vocoders (SMV). RFC Editor, 2003. http://dx.doi.org/10.17487/rfc3558.

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Heide, David A., Aaron E. Cohen, Yvette T. Lee, and Thomas M. Moran. Universal Vocoder Using Variable Data Rate Vocoding. Defense Technical Information Center, 2013. http://dx.doi.org/10.21236/ada588068.

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Mack, M. A., and B. Gold. The Intelligibility of Non-Vocoded and Vocoded Semantically Anomalous Sentences. Defense Technical Information Center, 1985. http://dx.doi.org/10.21236/ada160401.

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Mack, M., J. Tierney, and M. E. Boyle. The Intelligibility of Natural and LPC-Vocoded Words and Sentences Presented to Native and Non-Native Speakers of English. Defense Technical Information Center, 1990. http://dx.doi.org/10.21236/ada226180.

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