Bibliografías temáticas / Speech coding

Literatura académica sobre el tema "Speech coding"

Autor: Grafiati
Publicado: 4 de junio de 2021
Última modificación: 19 de febrero de 2023

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Artículos de revistas sobre el tema "Speech coding"

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Rao, P. Srinivasa, Ch ChandraLekha, B. Srinivasa Reddy, G. Lakshmi Deepthi y K. Bhargav. "Speech Coding". International Journal of Future Generation Communication and Networking 7, n.º 5 (31 de octubre de 2014): 229–38. http://dx.doi.org/10.14257/ijfgcn.2014.7.5.19.

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Yatsuzuka, Yohtaro. "Speech coding system". Journal of the Acoustical Society of America 90, n.º 1 (julio de 1991): 626. http://dx.doi.org/10.1121/1.402304.

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Sasaki, Seishi. "Speech coding circuit". Journal of the Acoustical Society of America 97, n.º 1 (enero de 1995): 736. http://dx.doi.org/10.1121/1.413056.

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Jagtap, S. K., M. S. Mulye y M. D. Uplane. "Speech Coding Techniques". Procedia Computer Science 49 (2015): 253–63. http://dx.doi.org/10.1016/j.procs.2015.04.251.

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Alsaka, Y. A. "Contractive speech coding". Electronics Letters 28, n.º 14 (1992): 1358. http://dx.doi.org/10.1049/el:19920863.

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Amano, Fumio. "Speech coding apparatus". Journal of the Acoustical Society of America 95, n.º 2 (febrero de 1994): 1183. http://dx.doi.org/10.1121/1.408451.

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Goldberg, Randy G. "Perpetual speech coding". Journal of the Acoustical Society of America 96, n.º 4 (octubre de 1994): 2595. http://dx.doi.org/10.1121/1.410069.

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Taniguchi, Tomohiko y Mark A. Johnson. "Speech coding system". Journal of the Acoustical Society of America 96, n.º 3 (septiembre de 1994): 1949. http://dx.doi.org/10.1121/1.410190.

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Taniguchi, Tomohiko. "Speech coding apparatus using multimode coding". Journal of the Acoustical Society of America 95, n.º 5 (mayo de 1994): 2795. http://dx.doi.org/10.1121/1.409797.

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ros, Fi A. y Utpal Bhattacharjee. "Impact of Semantic Coding of Emotional Speech on Speech Coding Performance". International Journal of Computer Trends and Technology 57, n.º 1 (25 de marzo de 2018): 6–10. http://dx.doi.org/10.14445/22312803/ijctt-v57p102.

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Tesis sobre el tema "Speech coding"

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Abboud, Karim. "Wideband CELP speech coding". Thesis, McGill University, 1992. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=56805.

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The purpose of this thesis is to study the coding of wideband speech and to improve on previous Code-Excited Linear Prediction (CELP) coders in terms of speech quality and bit rate. To accomplish this task, improved coding techniques are introduced and the operating bit rate is reduced while maintaining and even enhancing the speech quality.
the first approach considers the quantization of Liner Predictive Coding (LPC) parameters and uses a three way split vector quantization. Both scalar and vector quantization are initially studied; results show that, with adequate codebook training, the second method generates better results while using a fewer number of bits. Nevertheless, the use of vector quantizers remain highly complex in terms of memory and number of computations. A new quantization scheme, split vector quantization (split VQ), is investigated to overcome this complexity problem. Using a new weighted distance measure as a selection criterion for split VQ, the average spectral distortion is significantly reduced to match the results obtained with scalar quantizers.
The second approach introduces a new pitch predictor with an increased temporal resolution for periodicity. This new technique has the advantage of maintaining the same quality obtained with conventional multiple coefficient predictors at a reduced bit rate. Furthermore, the conventional CELP noise weighting filter is modified to allow more freedom and better accuracy in the modeling of both tilt and formant structures. Throughout this process, different noise weighting schemes are evaluated and the results show that the new filter greatly contributes in solving the problem of high frequency distortion.
The final wideband CELP coder is operational at 11.7 kbits/s and generates a high perceptual quality of the reconstructed speech using the fractional pitch predictor and the new perceptual noise weighting filter.
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Sturt, Christian. "Pitch synchronous speech coding techniques". Thesis, University of Surrey, 2003. http://epubs.surrey.ac.uk/843327/.

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Efficient source coding techniques are necessary to make optimal use of the limited bandwidth available in mobile phone networks. Most current mobile telephone communication systems compress the speech waveform by using speech coders based on the Code Excited Linear Prediction (CELP) model. Such coders give high quality speech at bit rates of 8 kbps and above. Below 8 kbps, the quality of the coded speech degrades rapidly. At rates of 6 kbps and below, parametric speech coders offer better speech quality. These coders reduce the required bit rate by transmitting certain characteristics of the speech waveform to the decoder, rather than attempting to code the waveform itself. The disadvantage of parametric coders is that the maximum achievable quality is limited by assumptions made during the coding of the speech signal. The aim of the research presented is to investigate and eliminate the factors that limit the speech quality of parametric coders. A new pitch synchronous coding model is proposed that operates on individual pitch cycle waveforms of speech rather than longer, fixed length frames as used in classic techniques. In order to implement a pitch synchronous coder, new pitch cycle detection algorithms have been proposed. Pitch synchronous parameter analysis was investigated and several new techniques have been developed. A novel pitch synchronous split-band voicing estimator has been proposed that utilises only the phase of the speech harmonics rather than the periodicity used in traditional techniques. Fixed rate quantisation of pitch synchronous speech parameters has been investigated and a joint quantisation/interpolation scheme has been proposed. This scheme has been applied to the quantisation of the pitch synchronous parameters and has been shown to outperform traditional quantisation techniques. A comparison of a reference parametric coder with its pitch synchronous counterpart has shown that the pitch synchronous paradigm eliminates some of the main factors that limit the speech quality in parametric coders. It is expected that this will lead to the development of speech coders that can produce speech of higher quality than current parametric coders operating at the same bit rate. Key words: Speech Coding, Pitch Synchronous, Sinusoidal Coding, Split-Band LPC Coding.
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Kaouri, Hussein Ali. "Speech coding using vector quantisation". Thesis, Queen's University Belfast, 1988. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.356934.

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Kritzinger, Carl. "Low bit rate speech coding". Thesis, Stellenbosch : University of Stellenbosch, 2006. http://hdl.handle.net/10019.1/2078.

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Thesis (MScIng (Electrical and Electronic Engineering))--University of Stellenbosch, 2006.
Despite enormous advances in digital communication, the voice is still the primary tool with which people exchange ideas. However, uncompressed digital speech tends to require prohibitively high data rates (upward of 64kbps), making it impractical for many applications. Speech coding is the process of reducing the data rate of digital voice to manageable levels. Parametric speech coders or vocoders utilise a-priori information about the mechanism by which speech is produced in order to achieve extremely efficient compression of speech signals (as low as 1 kbps). The greater part of this thesis comprises an investigation into parametric speech coding. This consisted of a review of the mathematical and heuristic tools used in parametric speech coding, as well as the implementation of an accepted standard algorithm for parametric voice coding. In order to examine avenues of improvement for the existing vocoders, we examined some of the mathematical structure underlying parametric speech coding. Following on from this, we developed a novel approach to parametric speech coding which obtained promising results under both objective and subjective evaluation. An additional contribution by this thesis was the comparative subjective evaluation of the effect of parametric speech coding on English and Xhosa speech. We investigated the performance of two different encoding algorithms on the two languages.
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Burnett, I. S. "Hybrid techniques for speech coding". Thesis, University of Bath, 1992. https://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.317353.

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Al-Naimi, Khaldoon Taha. "Advanced speech processing and coding techniques". Thesis, University of Surrey, 2002. http://epubs.surrey.ac.uk/843488/.

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Over the past two decades there has been substantial growth in speech communications and new speech related applications. Bandwidth constraints led researchers to investigate ways of compressing speech signals whilst maintaining speech quality and intelligibility so as to increase the possible number of customers for the given bandwidth. Because of this a variety of speech coding techniques have been proposed over this period. At the heart of any proposed speech coding method is quantisation of the speech production model parameters that need to be transmitted to the decoder. Quantisation is a controlling factor for the targeted bit rates and for meeting quality requirements. The objectives of the research presented in this thesis are twofold. The first enabling the development of a very low bit rate speech coder which maintains quality and intelligibility. This includes increasing the robustness to various operating conditions as well as enhancing the estimation and improving the quantisation of speech model parameters. The second objective is to provide a method for enhancing the performance of an existing speech related application. The first objective is tackled with the aid of three techniques. Firstly, various novel estimation techniques are proposed which are such that the resultant estimated speech production model parameters have less redundant information and are highly correlated. This leads to easier quantisation (due to higher correlation) and therefore to bit saving. The second approach is to make use of the joint effect of the quantisation of spectral parameters (i.e. LSF and spectral amplitudes) for their big impact on the overall bit allocation required. Work towards the first objective also includes a third technique which enhances the estimation of a speech model parameter (i.e. the pitch) through a robust statistics-based post-processing (or tracking) method which operates in noise contaminated environments. Work towards the second objective focuses on an application where speech plays an important role, namely echo-canceller and noise-suppressor systems. A novel echo-canceller method is proposed which resolves most of the weaknesses present in existing echo-canceller systems and improves the system performance.
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Zhao, David Yuheng. "Model Based Speech Enhancement and Coding". Doctoral thesis, Stockholm : Kungliga Tekniska högskolan, 2007. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-4412.

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Katugampala, Nilantha N. "Multimode speech coding below 6 kbps". Thesis, University of Surrey, 2001. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.365141.

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The past two decades have witnessed a rapid expansion of the telecommunications industry. This growth has been primarily fuelled by the proliferation of the digital communication systems and services which have become easily available through wired and wireless networks. Current research trends involving integration and packetisation of voice, video and data channels into true multimedia communications, promise a similar technological revolution in the next decade. The available bandwidth in wire based terrestrial network is a relatively cheap and expandable resource. However in satellite and cellular radio systems the bandwidth is inherently limited and an expensive resource. In order to accommodate ever growing numbers of subscribers whilst maintaining high quality and low operational costs, it is essential to maximise the spectral efficiency. The research presented in this thesis has focused on the development of new source compression algorithms, tailored for human speech in order to improve the spectral efficiency of digital transmission systems. Recently there is an increasing interest on speech coding algorithms which combine various existing technologies in order to improve the speech quality .whilst maintaining the low transmission rate of the existing coding techniques. The aim of the research presented in this thesis was to develop a complete hybrid coding algorithm which combines harmonic and waveform approximating coding techniques. In order to integrate the two coding paradigms novel phase synchronisation and classification techniques were developed. The perceptual quality of the speech synthesised using the unquantised hybrid model achieves nearly transparent quality. The hybrid model was used to develop variable bit rate coders, which are particularly advantageous for voice storage, Code Division Multiple Access (CDMA) wireless networks, packet switched networks, and statistical multiplexing of speech for multi channel communications.
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Green, Richard C. "Walsh based cepstra for speech coding". Thesis, King's College London (University of London), 1991. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.392848.

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Ooi, James M. 1970. "Application of wavelets to speech coding". Thesis, Massachusetts Institute of Technology, 1993. http://hdl.handle.net/1721.1/12340.

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Libros sobre el tema "Speech coding"

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Bäckström, Tom. Speech Coding. Cham: Springer International Publishing, 2017. http://dx.doi.org/10.1007/978-3-319-50204-5.

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Ogunfunmi, Tokunbo. Principles of speech coding. Boca Raton: Taylor & Francis, 2010.

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Ayuso, Antonio J. Rubio y Juan M. López Soler, eds. Speech Recognition and Coding. Berlin, Heidelberg: Springer Berlin Heidelberg, 1995. http://dx.doi.org/10.1007/978-3-642-57745-1.

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Atal, Bishnu S., Vladimir Cuperman y Allen Gersho, eds. Advances in Speech Coding. Boston, MA: Springer US, 1991. http://dx.doi.org/10.1007/978-1-4615-3266-8.

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Atal, Bishnu S., Vladimir Cuperman y Allen Gersho. Advances in speech coding. Editado por IEEE Workshop on Speech Coding for Telecommunications (1989 : Vancouver, B.C.). New York: Springer, 1991.

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S, Atal Bishnu, Cuperman Vladimir, Gersho Allen y IEEE Workshop on Speech Coding for Telecommunications (1989 : Vancouver, B.C.), eds. Advances in speech coding. Boston: Kluwer Academic Publishers, 1991.

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Papamichalis, Panos E. Practical approaches to speech coding. Englewood Cliffs, N.J: Prentice-Hall, 1987.

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Nejat, İnce A., ed. Digital speech processing: Speech coding, synthesis, and recognition. Boston: Kluwer Academic Publishers, 1991.

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İnce, A. Nejat. Digital Speech Processing: Speech Coding, Synthesis and Recognition. Boston, MA: Springer US, 1992.

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Barnwell, T. P. Speech coding: A computer laboratory textbook. New York: Wiley, 1996.

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Capítulos de libros sobre el tema "Speech coding"

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Gersho, Allen. "Speech Coding". En The Kluwer International Series in Engineering and Computer Science, 73–100. Boston, MA: Springer US, 1992. http://dx.doi.org/10.1007/978-1-4757-2148-5_3.

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Owens, F. J. "Speech Coding". En Signal Processing of Speech, 122–37. London: Macmillan Education UK, 1993. http://dx.doi.org/10.1007/978-1-349-22599-6_6.

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Moreau, Nicolas. "Speech Coding". En Tools for Signal Compression, 101–22. Hoboken, NJ USA: John Wiley & Sons, Inc., 2013. http://dx.doi.org/10.1002/9781118616611.ch6.

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Alencar, Marcelo S. y Valdemar C. da Rocha. "Speech Coding". En Communication Systems, 97–134. Cham: Springer International Publishing, 2022. http://dx.doi.org/10.1007/978-3-031-12067-1_3.

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Macario, R. C. V. "Speech Coding". En Cellular Radio, 158–71. London: Macmillan Education UK, 1997. http://dx.doi.org/10.1007/978-1-349-14433-4_7.

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Alencar, Marcelo S. y Valdemar C. da Rocha. "Speech Coding". En Communication Systems, 89–128. Cham: Springer International Publishing, 2019. http://dx.doi.org/10.1007/978-3-030-25462-9_3.

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Taniguchi, Tomohiko, Yoshinori Tanaka y Robert M. Gray. "Speech Coding with Dynamic Bit Allocation (Multimode Coding)". En Advances in Speech Coding, 157–66. Boston, MA: Springer US, 1991. http://dx.doi.org/10.1007/978-1-4615-3266-8_16.

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Gabor, G. y Z. Györfi. "Design Examples—Speech Compression". En Recursive Source Coding, 67–75. New York, NY: Springer New York, 1986. http://dx.doi.org/10.1007/978-1-4613-8649-0_4.

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Ohm, Jens. "Speech and Audio Coding". En Multimedia Signal Coding and Transmission, 463–90. Berlin, Heidelberg: Springer Berlin Heidelberg, 2015. http://dx.doi.org/10.1007/978-3-662-46691-9_8.

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Cuperman, Vladimir y Peter Lupini. "Variable Rate Speech Coding". En Modern Methods of Speech Processing, 101–20. Boston, MA: Springer US, 1995. http://dx.doi.org/10.1007/978-1-4615-2281-2_5.

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Actas de conferencias sobre el tema "Speech coding"

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CHENCHAMMA, G., P. L. CHOWDARY y K. SHALINI KATYAYANI. "SPEECH CODING WITH LINEAR PREDICTIVE CODING". En Proceedings of the International Conference on ICMEE 2009. WORLD SCIENTIFIC, 2009. http://dx.doi.org/10.1142/9789814289795_0001.

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Noll, Peter. "Speech coding for communications". En 3rd European Conference on Speech Communication and Technology (Eurospeech 1993). ISCA: ISCA, 1993. http://dx.doi.org/10.21437/eurospeech.1993-6.

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Switzer, Shane, Tim Anderson, Matthew Kabrisky, Steven K. Rogers y Bruce Suter. "Frequency domain speech coding". En 2nd International Conference on Spoken Language Processing (ICSLP 1992). ISCA: ISCA, 1992. http://dx.doi.org/10.21437/icslp.1992-401.

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Jayant, N. S., J. D. Johnston y Y. Shoham. "Coding of wideband speech". En 2nd European Conference on Speech Communication and Technology (Eurospeech 1991). ISCA: ISCA, 1991. http://dx.doi.org/10.21437/eurospeech.1991-97.

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Pattichis, Marios S. "Session TP6a: Speech coding". En 2009 Conference Record of the Forty-Third Asilomar Conference on Signals, Systems and Computers. IEEE, 2009. http://dx.doi.org/10.1109/acssc.2009.5469986.

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Lotfidereshgi, Reza y Philippe Gournay. "Cognitive Coding Of Speech". En ICASSP 2022 - 2022 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP). IEEE, 2022. http://dx.doi.org/10.1109/icassp43922.2022.9747914.

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Voran, Stephen y Andrew Catellier. "Multiple-Description Speech Coding Using Speech-Polarity Decomposition". En GLOBECOM 2010 - 2010 IEEE Global Communications Conference. IEEE, 2010. http://dx.doi.org/10.1109/glocom.2010.5683769.

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Chen, Juin-Hwey y Jes Thyssen. "The Broadvoice Speech Coding Algorithm". En 2007 IEEE International Conference on Acoustics, Speech, and Signal Processing. IEEE, 2007. http://dx.doi.org/10.1109/icassp.2007.366968.

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Sivaram, G. S. V. S., Sridhar Krishna Nemala, Mounya Elhilali, Trac D. Tran y Hynek Hermansky. "Sparse coding for speech recognition". En 2010 IEEE International Conference on Acoustics, Speech and Signal Processing. IEEE, 2010. http://dx.doi.org/10.1109/icassp.2010.5495649.

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Sehgal, Vivek Kumar, Shilpy Arora, Sahil Jain, Shantanu Agarwal, Ujjawal Khandelwal y Karan Jain. "Parameteric coding of speech signals". En Workshops (ICUMT). IEEE, 2009. http://dx.doi.org/10.1109/icumt.2009.5345381.

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Informes sobre el tema "Speech coding"

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Hogden, J. An articulatorily constrained, maximum entropy approach to speech recognition and speech coding. Office of Scientific and Technical Information (OSTI), diciembre de 1996. http://dx.doi.org/10.2172/432946.

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St. George, Brett A. Speech Coding and Phoneme Classification Using a Back-Propagation Neural Network. Fort Belvoir, VA: Defense Technical Information Center, mayo de 1997. http://dx.doi.org/10.21236/ada418472.

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Hogden, J. Improving on hidden Markov models: An articulatorily constrained, maximum likelihood approach to speech recognition and speech coding. Office of Scientific and Technical Information (OSTI), noviembre de 1996. http://dx.doi.org/10.2172/431136.

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Yesha, Yaacov. Channel coding for code excited linear prediction (CELP) encoded speech in mobile radio applications. Gaithersburg, MD: National Institute of Standards and Technology, 1994. http://dx.doi.org/10.6028/nist.ir.5503.

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Weiner, Andrew M. Spectral Methods for High-Speed Optical Transmultiplexing and Coding. Fort Belvoir, VA: Defense Technical Information Center, junio de 1998. http://dx.doi.org/10.21236/ada346411.

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Weiner, Andrew M. Femtosecond Laser System for Research on High-Speed Optical Transmultiplexing and Coding. Fort Belvoir, VA: Defense Technical Information Center, agosto de 1996. http://dx.doi.org/10.21236/ada323290.

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Weiner, Andrew M. Femtosecond Laser System for Research on High-Speed Optical Transmultiplexing and Coding. Fort Belvoir, VA: Defense Technical Information Center, septiembre de 1997. http://dx.doi.org/10.21236/ada343619.

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Weiner, Andrew M. Instrumentation for Research on High-Speed Optical Transmultiplexing and Coding for Optoelectronic Computer Networking. Fort Belvoir, VA: Defense Technical Information Center, abril de 1999. http://dx.doi.org/10.21236/ada363026.

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Dodge, M. y B. Ray. Definitions of Managed Object Extensions for Very High Speed Digital Subscriber Lines (VDSL) Using Single Carrier Modulation (SCM) Line Coding. RFC Editor, mayo de 2005. http://dx.doi.org/10.17487/rfc4069.

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Dodge, M. y B. Ray. Definitions of Managed Object Extensions for Very High Speed Digital Subscriber Lines (VDSL) Using Multiple Carrier Modulation (MCM) Line Coding. RFC Editor, mayo de 2005. http://dx.doi.org/10.17487/rfc4070.

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