Relevant bibliographies by topics / Speech Codes / Dissertations / Theses
To see the other types of publications on this topic, follow the link: Speech Codes.

Dissertations / Theses on the topic 'Speech Codes'

Author: Grafiati
Published: 4 June 2021
Last updated: 7 February 2022

Create a spot-on reference in APA, MLA, Chicago, Harvard, and other styles

Select a source type:

Consult the top 50 dissertations / theses for your research on the topic 'Speech Codes.'

Next to every source in the list of references, there is an 'Add to bibliography' button. Press on it, and we will generate automatically the bibliographic reference to the chosen work in the citation style you need: APA, MLA, Harvard, Chicago, Vancouver, etc.

You can also download the full text of the academic publication as pdf and read online its abstract whenever available in the metadata.

Browse dissertations / theses on a wide variety of disciplines and organise your bibliography correctly.

1

Johnson, Aaron Keith, and Aaron Keith Johnson. "Campus Speech Codes: A Legal and Philosophical View." Thesis, The University of Arizona, 2017. http://hdl.handle.net/10150/625015.

Full text
Abstract:
The goal of this paper is to determine what needs to change about campus speech codes in order for them to succeed against First Amendment challenges. Campus speech codes are a popular solution to the problem of hate speech on campuses. However, many commentators argue that these speech codes are either unethical or unconstitutional. Additionally, speech codes have historically been struck down the courts. This paper assesses the legal history of hate speech regulation, the commentary surrounding the law, and prior court cases in which speech codes were struck down in order to determine what types of hate speech are valid targets of regulation and why speech codes have been struck down in the past. Further, this paper attempts to determine what types of hate speech actually should be regulated based on ethical and practical considerations. Finally, this paper provides a set of guidelines which should help universities construct morally permissible speech codes which will succeed against First Amendment challenges.
APA, Harvard, Vancouver, ISO, and other styles
2

Rouchy, Christophe. "Systematic Design of Space-Time Convolutional Codes." Thesis, University of California, Santa Cruz, 2014. http://pqdtopen.proquest.com/#viewpdf?dispub=1554232.

Full text
Abstract:

Space-time convolutional code (STCC) is a technique that combines transmit diversity and coding to improve reliability in wireless fading channels. In this proposal, we demonstrate a systematic design of multi-level quadrature amplitude modulation (M-QAM) STCCs utilizing quadrature phase shift keying (QPSK) STCC as component codes for any number of transmit antennas. Morever, a low complexity decoding algorithm is introduced, where the decoding complexity increases linearly by the number of transmit antennas. The approach is based on utilizing a group interference cancellation technique also known as combined array processing (CAP) technique.

Finally, our research topic will explore: with the current approach, a scalable STTC with better performance as compared to space- time block code (STBC) combined with multiple trellis coded modulation (MTCM) also known as STBC-MTCM; the design of low complexity decoder for STTC; the combination of our approach with multiple-input multiple-output orthogonal frequency division multiplexing (MIMO-OFDM).

APA, Harvard, Vancouver, ISO, and other styles
3

Leighter, James L. "Codes of commonality and cooperation : notions of citizen personae and citizen speech codes in American public meetings /." Thesis, Connect to this title online; UW restricted, 2007. http://hdl.handle.net/1773/6178.

Full text
APA, Harvard, Vancouver, ISO, and other styles
4

Minter, Sam. "Speech on College Campuses: Methods, Motives, and Movements." Scholarship @ Claremont, 2017. http://scholarship.claremont.edu/cmc_theses/1698.

Full text
Abstract:
Are campus movements concerning free speech—from Berkeley in the 1960s to the campaign against political correctness today—really about speech? Are movements really concerned with civil liberties on campus or are their calls for free speech excited by partisan motives? While free speech movements are never purely driven by civil libertarian concerns, they should not be considered simply partisan either. Campus speech movements have frequently united activists across the ideological spectrum, which suggests that these movements aren’t only sectarian in nature. It also confirms that these movements are in fact about speech, because those advocating for it have a wide range of motives, but free speech is the point of agreement. However, this is not to say that there aren’t ulterior partisan underpinnings in these pushes for free speech.
APA, Harvard, Vancouver, ISO, and other styles
5

Rao, Sudha Suzanne. "Literacy as a learner variable in the use of salient letter codes for dedicated speech computers." Thesis, University of British Columbia, 1989. http://hdl.handle.net/2429/27622.

Full text
Abstract:
Literacy level is an important user variable in the process of selecting an appropriate augmentative communication device for a nonspeaking individual. This study investigated how much literacy was sufficient for a child to learn and remember salient letter codes to access prestored communicative messages from the memory of dedicated speech computers. Recent investigations (Light et al., 1988) have demonstrated that salient letter codes are the type of code most easily and accurately remembered by nonspeaking, literate adults. The present study examined the use of the salient letter encoding technique by children with varying degrees of literacy. The performance of three groups of normal children (19 literate, 21 emergent literate and 21 preliterate) was measured in terms of error rate and strategy use as a function of literacy ability after specific codes and the salient letter encoding strategy were explicitly taught for accessing ten communicative messages. Error analysis showed that the emergent literate and literate groups used the salient letter encoding strategy whereas the preliterate group used two ineffective visual strategies. Mean accuracy scores indicated mastery of the salient letter encoding technique by literate subjects (89% correct), sufficient performance by emergent literate subjects (66% correct) and very poor performance by preliterate subjects (27% correct). The accuracy scores and patterns of strategy use indicated that emergent literacy skills were sufficient for use of salient letter codes. It seems likely that future research using personalized codes with emergent literate children may demonstrate improved accuracy. The generalizability of these results to disordered populations and application to iconic systems was discussed. Extrapolated to the nonspeaking population, the results indicate that literate or emergent literate nonspeaking children would be capable users of salient letter codes. The performance of the three experimental groups was compared from the heuristic of a procedural view of memory with regard to opposing views of the nature of psycholinguistic and literacy development.
Medicine, Faculty of
Audiology and Speech Sciences, School of
Graduate
APA, Harvard, Vancouver, ISO, and other styles
6

Amazouz, Djegdjiga. "Linguistic and phonetic investigations of French-Algerian Arabic code-switching : large corpus studies using automatic speech processing." Thesis, Paris 3, 2019. http://www.theses.fr/2019PA030006.

Full text
Abstract:
Cette thèse présente des recherches linguistiques et phonétiques sur le code-switching Français-Arabe Algérien. Un corpus de 7h30 de parole (5h de parole spontané et 2h30 de parole lue) a été constitué en enregistrant 20 hommes et femmes parlant le français et l'arabe algérien. Cette thèse présente également les méthodes de traitement des données orales du code-switching telles que la segmentation de la parole, la segmentation des énoncés de code-switching ainsi que la transcription du français et du dialecte arabe algérien. Cette thèse présente également des méthodes d'alignement automatique de ces données bilingues ainsi qu'un alignement combiné de deux alignements monolingues. Nous avons mené des expériences basées sur l'alignement automatique avec des variations qui traitent de la question de l'influence d'un système phonologique d'une langue A sur des productions phonétiques en code-switching du français et de l'arabe algérien. Nous avons d'abord abordé la variation en réalisant une étude sur la variation des voyelles, dans des productions en langue française et en arabe algérien. Nous avons aussi abordé les consonnes emphatiques et l'emphatisation des deux langues. Enfin, nous avons également travaillé sur les géminées et la gémination dans les productions langagières en code-switching. Les résultats ont montré que le code-switching FR-AA se caractérise par des changements de langues très courts qui sont un réel défi pour l’identification des langues dans le code-switching. Le code-switching a un impact sur la variation phonétique des voyelles et des consonnes. La parole du code-switching permet au locuteur de produire moins de variation de voyelles et de consonnes que la parole monolingue
This thesis proposes linguistic and phonetic investigations of French-Algerian Arabic code-switching. A corpus of 7h30 of speech (5h of spontaneous speech and 2h30 of read speech) has been designed with 20 males and females French-Algerian Arabic speakers.This thesis also proposes code-switching speech data processing methods such as language segmentation, code-switching utterance segmentation and transcription of French and Algerian Arabic dialect. Automatic speech alignment methods of the code-switching data are proposed with combined alignment of two monolingual alignments. We conducted experiments based on language automatic identification and automatic alignment with variations that deals with the question of the influence of a phonological system of a language A on code-switching speech in phonetic productions of French and Algerian Arabic. We dealt first with identifying the language change boundaries. We performed also a variation study on vowel variation, in both French and Arabic productions. Finally, we dealt with three types of consonant variation in the code-switching speech: gemination, emphatization and voicing consonant as variants in production. The results shown that the code-switching French-Algerian Arabic is characterized by very short language switches witch constitute a big challenge to the code-switching languages identification . The code-switching has an impact of the phonetic variation in both vowel and consonants. The code-switching allows the speakers to produce less vowel and consonant variation than the monolingual speech
APA, Harvard, Vancouver, ISO, and other styles
7

How, Hee Thong. "Wideband speech and audio compression for wireless communications." Thesis, University of Southampton, 2001. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.342850.

Full text
APA, Harvard, Vancouver, ISO, and other styles
8

Madour, Lila. "A low-delay code excited linear prediction speech coder at 8 kbit/s /." Thesis, McGill University, 1994. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=68042.

Full text
Abstract:
The goal of this thesis is to design a high quality low-delay 8 kb/s speech coder. This research is motivated by the need of the telecommunication industries to standardize a high quality, low-delay and low rate speech coder. To meet these requirements, we use a coder based on code-existed linear prediction. To meet the demands of high quality and low bit rate, a vector quantizer is used to code the excitation signal. To meet the low-delay requirement, a backward adaptation technique of the synthesis filters is used. The focus of the research is on comparing different pitch synthesis filters in the CELP coder. From the three-order pitch synthesis filter, the first-order integer delay pitch synthesis filter and the first-order fractional delay pitch synthesis filter that are experimented in this research, the latter produces the best quality.
APA, Harvard, Vancouver, ISO, and other styles
9

Price, Moneca C. "Interactions between speech coders and disordered speech." Thesis, National Library of Canada = Bibliothèque nationale du Canada, 1997. http://www.collectionscanada.ca/obj/s4/f2/dsk2/ftp01/MQ28640.pdf.

Full text
APA, Harvard, Vancouver, ISO, and other styles
10

Bacha, Gabrielle Marie Bacha. "Individual and Community Rights Within University Conduct Systems." Ohio University Art and Sciences Honors Theses / OhioLINK, 2016. http://rave.ohiolink.edu/etdc/view?acc_num=ouashonors1461675735.

Full text
APA, Harvard, Vancouver, ISO, and other styles
11

Choy, Eddie L. T. "Waveform interpolation speech coder at 4 kbs." Thesis, McGill University, 1998. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=20901.

Full text
Abstract:
Speech coding at bit rates near 4 kbps is expected to be widely deployed in applications such as visual telephony, mobile and personal communications. This research focuses on developing a speech coder based on the waveform interpolation (WI) scheme, with an attempt to deliver near toll-quality speech at rates around 4 kbps. A WI coder has been simulated in floating-point using the C programming language. The high performance of the WI model has been confirmed by subjective listening tests in which the unquantized coder outperforms the 32 kbps G.726 standard (ADPCM) 98% of the time under clean input speech conditions; the reconstructed speech is perceived to be essentially indistinguishable from the original. When fully quantized, the speech quality of the WI coder at 4.25 kbps has been judged to be equivalent to or better than that of G.729 (the ITU-T toll-quality 8 kbps standard) for 45% of the test sentences. Further refinements of the quantization techniques are warranted to bring the coder closer to the toll-quality benchmark. Yet, the existing implementation has produced good quality coded speech with a high degree of intelligibility and naturalness when compared to the conventional coding schemes operating in the neighbourhood of 4 kbps.
APA, Harvard, Vancouver, ISO, and other styles
12

De, Aloknath. "Auditory distortion measures for speech coder evaluation." Thesis, McGill University, 1993. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=41270.

Full text
Abstract:
One of the important research problems in the area of speech coding is to determine the sound quality of coded speech signals. This quality can best be evaluated by a subjective assessment which is often difficult to administer and time consuming. An objective measure which is consistent with subjective assessment could play a vital role in the evaluation as well as in the design of a low bit-rate speech coder. In this dissertation, we introduce two distortion measures for speech coder evaluation. Since the perceptual abilities of a human being determine the precision with which speech data must be processed, we consider the details of cochlear (inner ear) and other auditory processing. Using Lyon's auditory model, the time-domain signal is mapped onto a perceptual-domain (PD). Any speech utterance is communicated to the brain through a series of all-or-none electrical spikes (firings) and the PD representation provides information pertaining to the probability-of-firings in the neural channels. Our first measure, namely the cochlear discrimination information (CDI), evaluates the cross-entropy of the neural firings for the coded speech with respect to those for the original one. With this measure, we also compute the rate-distortion function determining the lowest bit-rate required for a specified amount of distortion. In the second measure, namely the cochlear hidden Markovian (CHM) measure, we attempt to capture the high-level processing in the brain with simple hidden Markov models (HMMs). We characterize the firing events by HMMs where the order of occurrence of PD observations and correlations among adjacent observations are modeled suitably. For computing the coder distortion, the PD observations of the coded speech are matched against the HMMs derived from the PD observations of the original speech. Experimental results show that these measures conform to subjective evaluation results in majority of the cases. Finally, the introduced measures are also app
APA, Harvard, Vancouver, ISO, and other styles
13

Tyrberg, Andreas. "Data Transmission over Speech Coded Voice Channels." Thesis, Linköping University, Department of Electrical Engineering, 2006. http://urn.kb.se/resolve?urn=urn:nbn:se:liu:diva-6755.

Full text
Abstract:

The voice channel in mobile communication systems have high priority and are almost always available. By using the voice channel also for data transmissions it is possible to get the same availability as for voice calls. But due to speech codecs in the voice channel, regular modems can not be used and special techniques are needed to transmit data.

This thesis presents methods to transmit data over the voice channel in a GSM, UMTS or TETRA network. The focus has been on robust data transmission rather than high data bit rates. Approaches are introduced which improve the reliability for transmissions even for systems with low rate speech codecs and channels with some distortion.

The results of the thesis are suggestions of symbol patterns and ways to create and adapt symbols for specific application and channel conditions to achieve the desired goal for the application.

APA, Harvard, Vancouver, ISO, and other styles
14

Asenstorfer, John A. "Source-channel coding for CELP speech coders /." Title page, contents and abstract only, 1994. http://web4.library.adelaide.edu.au/theses/09PH/09pha816.pdf.

Full text
APA, Harvard, Vancouver, ISO, and other styles
15

Arabaci, Murat. "Nonbinary-LDPC-Coded Modulation Schemes for High-Speed Optical Communication Networks." Diss., The University of Arizona, 2010. http://hdl.handle.net/10150/195826.

Full text
Abstract:
IEEE has recently finished its ratification of the IEEE Standard 802.3ba in June 2010 which set the target Ethernet speed as 100 Gbps. The studies on the future trends of the ever-increasing demands for higher speed optical fiber communications show that there is no sign of decline in the demand. Constantly increasing internet traffic and the bandwidth-hungry multimedia services like HDTV, YouTube, voice-over-IP, etc. can be shown as the main driving forces. Indeed, the discussions over the future upgrades on the Ethernet speeds have already been initiated. It is predicted that the next upgrade will enable 400 Gbps Ethernet and the one after will be toward enabling the astounding 1 Tbps Ethernet.Although such high and ultra high transmission speeds are unprecedented over any transmission medium, the bottlenecks for achieving them over the optical fiber remains to be fundamental. At such high operating symbol rates, the signal impairments due to inter- and intra-channel fiber nonlinearities and polarization mode dispersion get exacerbated to the levels that cripple the high-fidelity communication over optical fibers. Therefore, efforts should be exerted to provide solutions that not only answer the need for high-speed transmission but also maintain low operating symbol rates.In this dissertation, we contribute to these efforts by proposing nonbinary-LDPC-coded modulation (NB-LDPC-CM) schemes as enabling technologies that can meet both the aforementioned goals. We show that our proposed NB-LDPC-CM schemes can outperform their prior-art, binary counterparts called bit-interleaved coded modulation (BI-LDPC-CM) schemes while attaining the same aggregate bit rates at a lower complexity and latency. We provide comprehensive analysis on the computational complexity of both schemes to justify our claims with solid evidence. We also compare the performances of both schemes by using amplified spontaneous emission (ASE) noise dominated optical fiber transmission and short to medium haul optical fiber transmission scenarios. Both applications show outstanding performances of NB-LDPC-CM schemes over the prior-art BI-LDPC-CM schemes with increasing gaps in coding gain as the transmission speeds increase. Furthermore, we present how a rate-adaptive NB-LDPC-CM can be employed to fully utilize the resources of a long haul optical transport network throughout its service time.
APA, Harvard, Vancouver, ISO, and other styles
16

Yatrou, Paul M. "Analysis of predictor mistracking in ADPCM speech coders." Thesis, McGill University, 1987. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=66242.

Full text
APA, Harvard, Vancouver, ISO, and other styles
17

Agarwal, Tarun. "Pre-processing of noisy speech for voice coders." Thesis, McGill University, 2002. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=33953.

Full text
Abstract:
Accurate Linear Prediction Coefficient (LPC) estimation is a central requirement in low bit-rate voice coding. Under harsh acoustic conditions, LPC estimation can become unreliable. This results in poor quality of encoded speech and introduces annoying artifacts.
The purpose of this thesis is to develop and test a two-branch speech enhancement pre-processing system. This system consists of two denoising blocks. One block will enhance the degraded speech for accurate LPC estimation. The second block will increase the perceptual quality of the speech to be coded. The goals of this research are two-fold---to design the second block, and to compare the performance of other denoising schemes in each of the two branches. Test results show that the two-branch system can provide better perceptual quality of coded speech over conventional one-branch (i.e., one denoising block) speech enhancement techniques under many noisy environments.
APA, Harvard, Vancouver, ISO, and other styles
18

Chen, Wei 1976. "Perceptual postfiltering for low bit rate speech coders." Thesis, McGill University, 2007. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=112563.

Full text
Abstract:
Adaptive postfiltering has become a common part of speech coding standards based on the Linear Prediction Analysis-by-Synthesis algorithm to decrease audible coding noise. However, a conventional adaptive postfilter is based on empirical assumptions of masking phenomena, which sometimes makes it hard to balance between noise reduction and speech distortion.
This thesis introduces a novel perceptual postfiltering system for low bit rate speech coders. The proposed postfilter works at the decoder, as is the case for the conventional adaptive postfilter. Specific human auditory properties are considered in the postfilter design to improve speech quality. A Gaussian Mixture Model based Minimum Mean Squared Error estimation of the perceptual postfilter is performed with the received information at the decoder. Perceptual postfiltering is then applied to the reconstructed speech to improve speech quality. Test results show that the proposed system gives better perceptual speech quality over conventional adaptive postfiltering.
APA, Harvard, Vancouver, ISO, and other styles
19

Choy, Eddie L. T. "Waveform interpolation speech coder at 4 kb/s." Thesis, National Library of Canada = Bibliothèque nationale du Canada, 1998. http://www.collectionscanada.ca/obj/s4/f2/dsk1/tape11/PQDD_0028/MQ50596.pdf.

Full text
APA, Harvard, Vancouver, ISO, and other styles
20

Neumeyer, Leonardo G. (Leonardo Gabriel) Carleton University Dissertation Engineering Electrical. "A low-delay backward-adaptive CELP speech coder." Ottawa, 1990.

Find full text
APA, Harvard, Vancouver, ISO, and other styles
21

Leung, Tze-Wo Carleton University Dissertation Engineering Systems and Computer. "Voice frame reconstruction methods for CELP speech coders." Ottawa, 1993.

Find full text
APA, Harvard, Vancouver, ISO, and other styles
22

Mustapha, Azhar K. 1975. "Postfiltering techniques in low bit-rate speech coders." Thesis, Massachusetts Institute of Technology, 1999. http://hdl.handle.net/1721.1/80589.

Full text
Abstract:
Thesis (M.Eng.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 1999.
Includes bibliographical references (leaves 78-80).
by Azhar K. Mustapha.
M.Eng.
APA, Harvard, Vancouver, ISO, and other styles
23

Hardwick, John C. (John Clark). "A 4.8 Kbps multi-band excitation speech coder." Thesis, Massachusetts Institute of Technology, 1988. http://hdl.handle.net/1721.1/14751.

Full text
APA, Harvard, Vancouver, ISO, and other styles
24

Brandstein, Michael Shapiro. "A 1.5 Kbps multi-band excitation speech coder." Thesis, Massachusetts Institute of Technology, 1990. http://hdl.handle.net/1721.1/14283.

Full text
Abstract:
Thesis (M.S.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 1990.
Includes bibliographical references (leaves 58-60).
by Michael Shapiro Brandstein.
M.S.
APA, Harvard, Vancouver, ISO, and other styles
25

Pereira, Wesley. "Modifying LPC parameter dynamics to improve speech coder efficiency." Thesis, McGill University, 2001. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=32970.

Full text
Abstract:
Reducing the transmission bandwidth and achieving higher speech quality are primary concerns in developing new speech coding algorithms. The goal of this thesis is to improve the perceptual speech quality of algorithms that employ linear predictive coding (LPC). Most LPC-based speech coders extract parameters representing an all-pole filter. This LPC analysis is performed on each block or frame of speech. To smooth out the evolution of the LPC tracks, each block is divided into subframes for which the LPC parameters are interpolated. This improves the perceptual quality without additional transmission bit rate. A method of modifying the interpolation endpoints to improve the spectral match over all the subframes is introduced. The spectral distortion and weighted Euclidean LSF (Line Spectral Frequencies) distance are used as objective measures of the performance of this warping method. The algorithm has been integrated in a floating point C-version of the Adaptive Multi Rate (AMR) speech coder and these results are presented.
APA, Harvard, Vancouver, ISO, and other styles
26

Konaté, Cheick Mohamed. "Enhancing speech coder quality: improved noise estimation for postfilters." Thesis, McGill University, 2011. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=104578.

Full text
Abstract:
ITU-T G.711.1 is a multirate wideband extension for the well-known ITU-T G.711 pulse code modulation of voice frequencies. The extended system is fully interoperable with the legacy narrowband one. In the case where the legacy G.711 is used to code a speech signal and G.711.1 is used to decode it, quantization noise may be audible. For this situation, the standard proposes an optional postfilter. The application of postfiltering requires an estimation of the quantization noise. The more accurate the estimate of the quantization noise is, the better the performance of the postfilter can be.In this thesis, we propose an improved noise estimator for the postfilter proposed for the G.711.1 codec and assess its performance. The proposed estimator provides a more accurate estimate of the noise with the same computational complexity.
ITU-T G.711.1 est une extension multi-débit pour signaux à large-bande de la très répandue norme de compression audio de UIT-T G.711. Cette extension est interoperationelle avec sa version initiale à bande étroite. Lorsque l'ancienne version G.711 est employée pour coder un signal vocal et que G.711.1 est utiliser pour le décoder, le bruit de quantificationpeut être entendu. Pour ce cas, la norme propose un post-filtre optionel. Le post-filtre nécessite l'estimation du bruit de quantification. La précision de l'estimation du bruit de quantification va jouer sur la performance du post-filtre.Dans cette thèse, nous proposons un meilleur estimateur du bruit de quantification pour le post-filtre proposé pour le codec G.711.1 et nous évaluons ses performances. L'estimateur que nous proposons donne une estimation plus précise du bruit de quantification avec la même complexité.
APA, Harvard, Vancouver, ISO, and other styles
27

Nagaswamy, Sriram. "Comparison of CELP speech coder with a wavelet method." Lexington, Ky. : [University of Kentucky Libraries], 2005. http://lib.uky.edu/ETD/ukyelen2006t00376/Thesis.pdf.

Full text
Abstract:
Thesis (M.S.)--University of Kentucky, 2005.
Title from document title page (viewed on January 30, 2006). Document formatted into pages; contains: ix, 124 p. : ill. Includes abstract and vita. Includes bibliographical references (p. 118-123).
APA, Harvard, Vancouver, ISO, and other styles
28

Papacostantinou, Costantinos. "Improved pitch modelling for low bit-rate speech coders." Thesis, National Library of Canada = Bibliothèque nationale du Canada, 1997. http://www.collectionscanada.ca/obj/s4/f2/dsk2/ftp01/MQ37279.pdf.

Full text
APA, Harvard, Vancouver, ISO, and other styles
29

Nagaswamy, Sriram. "Comparison of CELP speech coder with a wavelet method." UKnowledge, 2006. http://uknowledge.uky.edu/gradschool_theses/269.

Full text
Abstract:
This thesis compares the speech quality of Code Excited Linear Predictor (CELP, Federal Standard 1016) speech coder with a new wavelet method to compress speech. The performances of both are compared by performing subjective listening tests. The test signals used are clean signals (i.e. with no background noise), speech signals with room noise and speech signals with artificial noise added. Results indicate that for clean signals and signals with predominantly voiced components the CELP standard performs better than the wavelet method but for signals with room noise the wavelet method performs much better than the CELP. For signals with artificial noise added, the results are mixed depending on the level of artificial noise added with CELP performing better for low level noise added signals and the wavelet method performing better for higher noise levels.
APA, Harvard, Vancouver, ISO, and other styles
30

Wong, Wing-Tak Kenneth. "A speech coder design for land mobile radio communications." Thesis, University of Liverpool, 1989. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.237531.

Full text
APA, Harvard, Vancouver, ISO, and other styles
31

Nakhai, Mohammad Reza. "A low bit rate speech codec for wireless applications." Thesis, King's College London (University of London), 2000. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.392144.

Full text
APA, Harvard, Vancouver, ISO, and other styles
32

ARAUJO, ANTONIO MARCOS DE LIMA. "ANALYSIS OF WAVEFORM CODERS FOR SPEECH AND DATA SIGNALS." PONTIFÍCIA UNIVERSIDADE CATÓLICA DO RIO DE JANEIRO, 1986. http://www.maxwell.vrac.puc-rio.br/Busca_etds.php?strSecao=resultado&nrSeq=9246@1.

Full text
Abstract:
O trabalho examina o comportamento de Codificadores de forma de onda operando a 32,56 e 64kbit/s para transmissão digital de sinais de voz e de sinais de dados PSK-8 a 4800 bit/s e QAM-16 a 9600 bit/s. A partir de uma análise detalhada dos diversos sistemas, tanto em canal ideal como um canal ruidoso, é verificada a necessidade de se fazer uma identificação do tipo de sinal. De modo a permitir sua codificação de forma mais eficiente. É, então, proposta e avaliada a utilização de uma técnica de identificação estatística de sinais de voz e dados, em codificadores de forma de onda. A incorporação desta técnica ao sistema ADPCM a 32 kbit/s recomendado pelo CCITT permite uma melhoria do desempenho para sinais de dados, sem com isso alterar sua eficiência para sinais de voz.
This thesis evaluates the performance of waveform coders at 32,56 and 64kbit/s for digital transmission of speech signal and 4800 bit/s PSK-8 and 9600 bit/s QAM-16 voiceband data signas. A detailed analysis of the systems is carried out both under ideal and noisy channel conditions. From this analysis it was found that a scheme which accurately distinguishes the two classes of signals, would allow a more efficient encoding procedure. A method of statistical identification of speech and data signals is proposed and its use in wakeform coders is, then, analysed. The incorporation of this method into the 32 kbit/s ADPCM system recommended by CCITT provides an improvement in performance for data signals, without sacrificing its efficiency for speech signal.
APA, Harvard, Vancouver, ISO, and other styles
33

Ertan, Ali Erdem. "Pitch-synchronous processing of speech signal for improving the quality of low bit rate speech coders." Diss., Georgia Institute of Technology, 2004. http://hdl.handle.net/1853/36534.

Full text
APA, Harvard, Vancouver, ISO, and other styles
34

Ertan, Ali Erdem. "Pitch-synchronous processing of speech signal for improving the quality of low bit rate speech coders." Available online, Georgia Institute of Technology, 2004:, 2003. http://etd.gatech.edu/theses/available/etd-06072004-131138/unrestricted/ertan%5Fali%5Fe%5F200405%5Fphd.pdf.

Full text
Abstract:
Thesis (Ph. D.)--School of Electrical and Computer Engineering, Georgia Institute of Technology, 2004. Directed by Thomas P. Barnwell, III.
Vita. Includes bibliographical references (leaves 221-226).
APA, Harvard, Vancouver, ISO, and other styles
35

Menacer, Mohamed Amine. "Reconnaissance et traduction automatique de la parole de vidéos arabes et dialectales." Electronic Thesis or Diss., Université de Lorraine, 2020. http://www.theses.fr/2020LORR0157.

Full text
Abstract:
Les travaux de recherche ont été développés dans le cadre du projet AMIS (Access to Multilingual Information and opinionS) dont l'objectif principal est de développer un système d’aide à la compréhension de vidéos dans des langues étrangères en générant un résumé automatique de ces dernières dans une langue compréhensible par l'utilisateur. Dans le cadre de cette thèse, nous nous sommes concentrés sur la reconnaissance et la traduction automatique de la parole de vidéos arabes et dialectales. Les approches statistiques proposées dans la littérature pour la reconnaissance automatique de la parole (RAP) sont indépendantes de la langue et elles sont applicables à l'arabe standard. Cependant, cette dernière présente quelques caractéristiques que nous devons prendre en considération afin de booster les performances du système de RAP. Parmi ces caractéristiques on peut citer l'absence de l'indication des voyelles dans le texte ce qui rend difficile leur apprentissage par le modèle acoustique. Nous avons proposé plusieurs approches de modélisation acoustique et/ou de langage afin de mieux reconnaître la parole arabe. L'arabe standard n'est pas la langue maternelle, c'est pourquoi dans les conversations quotidiennes, on utilise le dialecte, un arabe inspiré de l'arabe standard, mais pas seulement. Nous avons travaillé sur l'adaptation du système développé pour l'arabe standard au dialecte algérien qui est l'une des variantes de la langue arabe les plus difficiles à reconnaître par les systèmes de RAP. Cela est dû aux mots empruntés d'autres langues, au code-switching et au manque de ressources. Notre proposition pour remédier à ces problèmes est de tirer profit des données orales et textuelles d'autres langues impactant le dialecte. Le texte résultant de la RAP arabe a été utilisé pour la traduction automatique (TA). Nous avons réalisé dans un premier temps une étude comparative entre l'approche statistique à base de segments et l'approche neuronale utilisées dans le cadre de la TA. Ensuite, nous nous sommes intéressés à l’adaptation de ces deux approches pour traduire le texte code-switché. Notre étude portait sur le mélange de l'arabe et de l'anglais dans des documents officiels des nations unies. Pour pallier les différents problèmes dus à la propagation des erreurs dans le système séquentiel, nous avons travaillé sur l'adaptation du vocabulaire du système de RAP et sur la proposition d'une nouvelle modélisation permettant la traduction directe de la parole
This research has been developed in the framework of the project AMIS (Access to Multilingual Information and opinionS). AMIS is an European project which aims to help people to understand the main idea of a video in a foreign language by generating an automatic summary of it. In this thesis, we focus on the automatic recognition and translation of the speech of Arabic and dialectal videos. The statistical approaches proposed in the literature for automatic speech recognition are language independent and they are applicable to modern standard Arabic. However, this language presents some characteristics that we need to take into consideration in order to boost the performance of the speech recognition system. Among these characteristics we can mention the absence of short vowels in the text, which makes their training by the acoustic model difficult. We proposed several approaches to acoustic and/or language modeling in order to better recognize the Arabic speech. In the Arab world, modern standard Arabic is not the mother tongue, that is why daily conversations are carried out with dialect, an Arabic inspired from modern standard Arabic, but not only. We worked on the adaptation of the speech recognition system developed for the modern standard Arabic to the Algerian dialect, which is one of the most difficult variants of the Arabic language to recognize by automatic speech recognition systems. This is mainly due to the borrowed words from other languages, the code-switching and the lack of resources. Our approach to overcome all these problems is to take advantage from oral and textual data of other languages that have an impact on the dialect in order to train the required models for dialect speech recognition. The resulting text from Arabic speech recognition system was then used for machine translation. As a starting point, we conducted a comparative study between the phrase based approach and the neural approach used in machine translation. Then, we adapted these two approaches to translate the code-switched text. Our study focused on the mix of Arabic and English in a parallel corpus extracted from official documents of the United Nations. In order to prevent the error propagation in the pipeline system, we worked on the adaptation of the vocabulary of the automatic speech recognition system and on the proposition of a new model that directly transforms a speech signal in language A into a sequence of words in another language B
APA, Harvard, Vancouver, ISO, and other styles
36

Woodard, Jason Paul. "Digital coding of speech using Code Excited Linear Prediction." Thesis, University of Southampton, 1995. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.387898.

Full text
APA, Harvard, Vancouver, ISO, and other styles
37

Khan, Mohammad M. A. "Coding of excitation signals in a waveform interpolation speech coder." Thesis, McGill University, 2001. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=32961.

Full text
Abstract:
The goal of this thesis is to improve the quality of the Waveform Interpolation (WI) coded speech at 4.25 kbps. The quality improvement is focused on the efficient coding scheme of voiced speech segments, while keeping the basic coding format intact. In the WI paradigm voiced speech is modelled as a concatenation of the Slowly Evolving pitch-cycle Waveforms (SEW). Vector quantization is the optimal approach to encode the SEW magnitude at low bit rates, but its complexity imposes a formidable barrier.
Product code vector quantizers (PC-VQ) are a family of structured VQs that circumvent the complexity obstacle. The performance of product code VQs can be traded off against their storage and encoding complexity. This thesis introduces split/shape-gain VQ---a hybrid product code VQ, as an approach to quantize the SEW magnitude. The amplitude spectrum of the SEW is split into three non-overlapping subbands. The gains of the three subbands form the gain vector which are quantized using the conventional Generalized Lloyd Algorithm (GLA). Each shape vector obtained by normalizing each subband by its corresponding coded gain is quantized using a dimension conversion VQ along with a perceptually based bit allocation strategy and a perceptually weighted distortion measure. At the receiver, the discontinuity of the gain contour at the boundary of subbands introduces buzziness in the reconstructed speech. This problem is tackled by smoothing the gain versus frequency contour using a piecewise monotonic cubic interpolant. Simulation results indicate that the new method improves speech quality significantly.
The necessity of SEW phase information in the WI coder is also investigated in this thesis. Informal subjective test results demonstrate that transmission of SEW magnitude encoded by split/shape-gain VQ and inclusion of a fixed phase spectrum drawn from a voiced segment of a high-pitched male speaker obviates the need to send phase information.
APA, Harvard, Vancouver, ISO, and other styles
38

Iyengar, Vasu. "A low delay 16 kbit/sec coder for speech signals /." Thesis, McGill University, 1987. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=63799.

Full text
APA, Harvard, Vancouver, ISO, and other styles
39

Zad-Issa, Mohammad R. "Smoothing the evolution of the spectral parameters in speech coders." Thesis, McGill University, 1998. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=20526.

Full text
Abstract:
The goal of this thesis is to modify the traditional linear prediction (LP) analysis in such way that the fluctuations of the LP coefficients are reduced, while the pitch pulse shape evolves slowly. These modifications can lead to an increase in the coding efficiency and/or an improvement in the speech quality. Two different methods have been developed for this purpose. In the first approach we derive the LP parameters such that the glottal excitation model matches as closely as possible a target waveform. The latter contains slowly evolving pulses representing voiced speech excitation. The simulation results indicate that the target matching method results in an increase in the pitch prediction gain which is a measure of similarity of successive pitch pulses. The frame-to-frame variation of the LP coefficients is also lowered with respect to the conventional linear prediction analysis. In the second method, we enforce the smoothness on the evolution of LP parameters by directly including their variation in the LP error function. A novel scheme to dynamically control the contribution of this additional term is also proposed. Experiments indicate that this method can considerably reduce the fluctuation of LP parameters while the overall prediction gain of the LP filter is maintained. (Abstract shortened by UMI.)
APA, Harvard, Vancouver, ISO, and other styles
40

Khan, Abdul Hannan. "Tree encoding in the ITU-T G.711.1 speech coder." Thesis, McGill University, 2011. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=97215.

Full text
Abstract:
This thesis examines further enhancement to ITU-T G.711.1 speech coder. The original G.711 coder is effectively a low band μ-law quantizer. The G.711.1 extension adds noise feed-back and lower band enhancement layer apart from the higher-band. To further improve the core lower-band coding performance the use of both vector quantization and delayed decision multi-path tree encoder in the above coder at the low band portion is studied. The delayed decision multi-path tree encoding is implemented by the (M,L) – algorithm. The new quantizer takes into account past history, and hence, the error propagation due to noise feed-back, and codes multiple samples under μ-law. The final bitstream is compatible with the G.711.1 decoder and, hence, with the original G.711 decoder. An evaluation method, ITU-T P.862 perceptual evaluation of speech quality (PESQ), is used to evaluate the performance. Both the vector quantizer and tree encoder have better performance than the original core layer encoder in terms of perceptual quality, though they are limited by the increased computational complexity. Future studies are suggested.
Cette thèse étudie en détail les améliorations apportées au codeur de la parole ITU-T G.711.1. Le codeur original G.711 est en fait un quantificateur μ-law. Le prolongement large-bande G.711.1 utilise le façonnage du bruit ainsi qu'une couche d'amélioration de la bande-basse en plus de la bande-haute. Afin d'améliorer le codage de la bande-basse principale, nous étudions l'utilisation de quantification vectorielle et la décision à retardement. Le codeur arboriforme avec décision à retardée est réalisé par l'algorithme(M,L). Le nouveau quantificateur considère l'information passée et par conséquent, il considère également la propagation de l'erreur engendrée par le façonnage du bruit. Il code plusieurs échantillons par μ-law. Le flot binaire final est compatible avec le décodeur du prolongement large-bande G.711.1 et donc naturellement avec le décodeur du G.711 original. Une méthode d'évaluation, ITU-T P.862 (PESQ) est utilisée pour évaluer la performance. Les résultats montrent que la quantification vectorielle et le codeur arboriforme sont perceptuellement plus performants que le codeur original de la bande principale. Nous notons tout de même qu'ils sont numériquement plus complexes à réaliser. Des études supplémentaires sont suggérées.
APA, Harvard, Vancouver, ISO, and other styles
41

Zad-Issa, Mohammad Reza. "Smoothing the evolution of the spectral parameters in speech coders." Thesis, National Library of Canada = Bibliothèque nationale du Canada, 1998. http://www.collectionscanada.ca/obj/s4/f2/dsk1/tape11/PQDD_0002/MQ44049.pdf.

Full text
APA, Harvard, Vancouver, ISO, and other styles
42

Edwards, Richard. "Advanced signal processing techniques for pitch synchronous sinusoidal speech coders." Thesis, University of Surrey, 2007. http://epubs.surrey.ac.uk/833/.

Full text
Abstract:
Recent trends in commercial and consumer demand have led to the increasing use of multimedia applications in mobile and Internet telephony. Although audio, video and data communications are becoming more prevalent, a major application is and will remain the transmission of speech. Speech coding techniques suited to these new trends must be developed, not only to provide high quality speech communication but also to minimise the required bandwidth for speech, so as to maximise that available for the new audio, video and data services. The majority of current speech coders employed in mobile and Internet applications employ a Code Excited Linear Prediction (CELP) model. These coders attempt to reproduce the input speech signal and can produce high quality synthetic speech at bit rates above 8 kbps. Sinusoidal speech coders tend to dominate at rates below 6 kbps but due to limitations in the sinusoidal speech coding model, their synthetic speech quality cannot be significantly improved even if their bit rate is increased. Recent developments have seen the emergence and application of Pitch Synchronous (PS) speech coding techniques to these coders in order to remove the limitations of the sinusoidal speech coding model. The aim of the research presented in this thesis is to investigate and eliminate the factors that limit the quality of the synthetic speech produced by PS sinusoidal coders. In order to achieve this innovative signal processing techniques have been developed. New parameter analysis and quantisation techniques have been produced which overcome many of the problems associated with applying PS techniques to sinusoidal coders. In sinusoidal based coders, two of the most important elements are the correct formulation of pitch and voicing values from the' input speech. The techniques introduced here have greatly improved these calculations resulting in a higher quality PS sinusoidal speech coder than was previously available. A new quantisation method which is able to reduce the distortion from quantising speech spectral information has also been developed. When these new techniques are utilised they effectively raise the synthetic speech quality of sinusoidal coders to a level comparable to that produced by CELP based schemes, making PS sinusoidal coders a promising alternative at low to medium bit rates.
APA, Harvard, Vancouver, ISO, and other styles
43

Lisk, Durodami J. (Durodami Joscelyn) 1976. "Transcoding between QCELP 13K and G.723.1 CELP speech coders." Thesis, Massachusetts Institute of Technology, 1999. http://hdl.handle.net/1721.1/80104.

Full text
Abstract:
Thesis (S.B. and M.Eng.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 1999.
Includes bibliographical references (leaves 77-78).
by Durodami J. Lisk.
S.B.and M.Eng.
APA, Harvard, Vancouver, ISO, and other styles
44

Chibani, Mohamed. "Increasing the Robustness of CELP Speech Codecs against packet losses." Thèse, Université de Sherbrooke, 2007. http://savoirs.usherbrooke.ca/handle/11143/1805.

Full text
Abstract:
Voice communication systems are moving at a steady pace from circuit-switched to packet-switched networks. Converged networks, in which data traffic and multimedia traffic share the same infrastructures, have yet to face many challenges. Unlike data traffic, voice traffic is very sensitive to issues such as delay, jitter and packet losses. The Quality of Service (QoS) of IP networks has greatly improved over the recent years. Nevertheless, it is not uncommon to experience loss in QoS on some networks caused by lack of resources or software/hardware failures. The speech codec (COder DECoder) is an essential component in a digital voice communication system. It converts the speech signal into a compact representation in order to efficiently use the available bandwidth. To compress the signal, speech codecs usually exploit the high redundancy present in a speech signal using techniques known as predictive coding. The most widely used speech coding model is the Code Excited Linear Prediction (CELP) model upon which are based most of the modern standardized speech codecs. The use of predictive codebooks, the Adaptive CodeBook (ACB) in particular, allows a tremendous reduction in bitrate. Unfortunately, the extensive use of prediction renders CELP codecs more vulnerable to packet losses than non-predictive codecs. A possible solution is to avoid using prediction at the cost of an increased bitrate. In this thesis, we propose several techniques to reduce the vulnerability of CELP codecs to packet losses caused by the use of prediction. Solutions that aim at limiting the contribution of the ACB and thus that reduce the sensitivity of the codec to packet losses are presented and discussed. Techniques based on sending side information, in order to help the decoder recover after a packet loss, are also proposed. The solutions are constrained to be low delay and low bitrate solutions, and are required to maintain interoperability if used in a standardized codec. Objective and subjective test results that demonstrate the performances of the proposed solutions are presented. We show that the robustness of CELP codecs can significantly be improved without sacrificing their compression performances. In spite of packet loss issues, we show that it is still beneficial to use predictive coding to efficiently exploit the available bandwidth.
APA, Harvard, Vancouver, ISO, and other styles
45

Gnaedig, David. "High-Speed decoding of convolutional Turbo Codes." Lorient, 2005. http://www.theses.fr/2005LORIS050.

Full text
Abstract:
Les turbocodes sont des codes obtenus par une concaténation de plusieurs codes convolutifs séparés par des entrelaceurs. En 1993, ils ont révolutionné le domaine du codage correcteur d’erreurs en s’approchant à quelques dixièmes de décibels de la limite théorique de Shannon. Ces performances sont d'autant plus remarquables que le principe itératif permet d'en effectuer le décodage avec une complexité matérielle limitée. Le succès des turbocodes s'est traduit par leur introduction dans plusieurs standards de communication. Les besoins croissants dans le domaine des réseaux large bande, nécessitent des implantations hauts débits qui posent de nouvelles problématiques L'objectif de cette thèse est d'étudier des architectures de décodage à haut débit offrant le meilleur compromis en terme de débit sur complexité. Dans un premier temps, nous avons proposé un modèle simple permettant d'exprimer le débit et l'efficacité d'une architecture. Ce modèle appliqué au turbo­ décodage met en évidence trois paramètres caractéristiques ayant un impact sur le débit et l'efficacité du décodeur : le degré de parallélisme, le taux d'utilisation (activité) des unités de calcul cl la fréquence d'horloge. Nous avons abordé chacun de ces points en explorant un large spectre de possibilités de l'espace de conception allant de la construction conjointe du code et du décodeur à l'optimisation directe des architectures de décodage pour un code ou un ensemble de codes prédéfinis. Nous avons tout d'abord proposé un nouveau schéma de codage appelé turbocodes à roulettes permettant de minimiser la memoire du décodeur par un décodage en parallèle d'un mot de code reçu par plusieurs processeurs à entrée et sortie souples. Afin de résoudre le problème des accès concurrents aux mémoires qui en résulte, nous avons conçu un nouvel entrelaceur hiérarchique. Nous avons ensuite exploré plusieurs solutions permettant d'améliorer l'activité des processeurs utilisation d'une architecture hybride série/parallèle et proposition de nouveaux séquencements au niveau interne des processeurs, et aussi au niveau global en association avec la construction d'entrelaceurs contraints adaptés. Enfin grace à méthode originale de réduction du chemin critique du calcul récursif des métriques de nœuds, nous avons obtenu, sans coût matériel supplémentaire pour un circuit FPGA, un doublement de la fréquence d'horloge du décodeur. La plupart des techniques développées dans cette thèse ont été validées par la réalisation d'un turbo-décodeur pour le standard d'accès sans-fil large bande WiMAX (IEEE 802. 16) qui atteint des performances de correction d'erreur excellentes pour un débit atteignant 100 Mbit/s sur un seul circuit FPGA
Turbo codes are built as a concatenation of several convolutional codes separated by interleavers. In 1993, they have revolutionized error correcting coding by approaching within a few tenths of a decibel the Shannon limit. This performance is even more astonishing because the iterative decoding principle enables the decoder to be implemented in hardware with a relative low complexity. Due to their success, they are now widely used in practical systems and open standards. The increasing demand for high throughput applications in broadband applications is strong)y calling for high-speed decoder implementations, thus leading to new challenges. The objective of this thesis is to study high-throughput decoding architectures offering the best throughput versus complexity trade-off. We first laid down a simple expression to evaluate the benefits of an architecture in terms of throughput and efficiency. The application of this model to turbo decoding highlighted three typical parameters influencing the throughput and efficiency of the decoder : the degree of parallelism, the ratio of utilization (activity) of the processing units and the clock frequency. We tackled each of these points by investigating a large spectrum of possibilities of the design space, ranging from the joint code and decoder design to the optimization of the decoder architecture for a given code or set of codes. We first proposed a new coding scheme called Multiple Slice Turbo Codes making possible to minimize the memory requirements of the decoder using the parallel decoding of a the received codeword by several soft-input soft-output processors. In order to solve the resulting concurrent accesses to the memory, we designed a novel hierarchical interleaver. Second, we explored several solutions for improving the activity of the processors including the usage of a hybrid parallel/serial architecture and the introduction of two new schedules for parallel decoding: one schedule internal to the processors, and another at a more global level in association with an adapted constrained interleaver. Finally, thanks to an original method to reduce the critical path in the recursive computation of state metrics, we obtained, at no cost on a FPGA circuit, a doubling of the maximal clock frequency of the decoder. Most of the w techniques developed in this thesis were validated by designing a turbo decoder for the wireless broadband access standard WiMAX (IEEE 802. 16) that achieves excellent error decoding performance reaching a throughput of 100Mbit/s on a single FPGA
APA, Harvard, Vancouver, ISO, and other styles
46

Batshon, Hussam George. "Coded Modulation for High Speed Optical Transport Networks." Diss., The University of Arizona, 2010. http://hdl.handle.net/10150/194075.

Full text
Abstract:
At a time where almost 1.75 billion people around the world use the Internet on a regular basis, optical communication over optical fibers that is used in long distance and high demand applications has to be capable of providing higher communication speed and re-liability. In recent years, strong demand is driving the dense wavelength division multip-lexing network upgrade from 10 Gb/s per channel to more spectrally-efficient 40 Gb/s or 100 Gb/s per wavelength channel, and beyond. The 100 Gb/s Ethernet is currently under standardization, and in a couple of years 1 Tb/s Ethernet is going to be standardized as well for different applications, such as the local area networks (LANs) and the wide area networks (WANs). The major concern about such high data rates is the degradation in the signal quality due to linear and non-linear impairments, in particular polarization mode dispersion (PMD) and intrachannel nonlinearities. Moreover, the higher speed transceivers are expensive, so the alternative approaches of achieving the required rates is preferably done using commercially available components operating at lower speeds.In this dissertation, different LDPC-coded modulation techniques are presented to offer a higher spectral efficiency and/or power efficiency, in addition to offering aggregate rates that can go up to 1Tb/s per wavelength. These modulation formats are based on the bit-interleaved coded modulation (BICM) and include: (i) three-dimensional LDPC-coded modulation using hybrid direct and coherent detection, (ii) multidimensional LDPC-coded modulation, (iii) subcarrier-multiplexed four-dimensional LDPC-coded modulation, (iv) hybrid subcarrier/amplitude/phase/polarization LDPC-coded modulation, and (v) iterative polar quantization based LDPC-coded modulation.
APA, Harvard, Vancouver, ISO, and other styles
47

Gould, Jonathan B. "Symbolic speech : legal mobilization and the rise of collegiate hate speech codes /." 1999. http://gateway.proquest.com/openurl?url_ver=Z39.88-2004&res_dat=xri:pqdiss&rft_val_fmt=info:ofi/fmt:kev:mtx:dissertation&rft_dat=xri:pqdiss:9920143.

Full text
APA, Harvard, Vancouver, ISO, and other styles
48

Culver, Kathleen Bartzen. "Campus hate speech codes and hostile environment sexual harassment law." 1992. http://catalog.hathitrust.org/api/volumes/oclc/28540192.html.

Full text
Abstract:
Thesis (M.A.)--University of Wisconsin--Madison, 1992.
Typescript. eContent provider-neutral record in process. Description based on print version record. Includes bibliographical references (leaves 130-136).
APA, Harvard, Vancouver, ISO, and other styles
49

YAN, BAI-HONG, and 顏百宏. "Recognition of continuous Mandarin speech using formant frequency location codes." Thesis, 1989. http://ndltd.ncl.edu.tw/handle/97950173344292981116.

Full text
APA, Harvard, Vancouver, ISO, and other styles
50

Asenstorfer, John A. (John Anthony). "Source-channel coding for CELP speech coders / J.A. Asenstorfer." 1994. http://hdl.handle.net/2440/18512.

Full text
Abstract:
Bibliography: leaves 197-205.
xiv, 205 leaves : ill. ; 30 cm.
Title page, contents and abstract only. The complete thesis in print form is available from the University Library.
This thesis is concerned with methods for protecting speech coding parameters transmitted over noisy channels. A linear prediction (LP) coder is employed to remove the short term correlations of speech. Protection of two sets of parameters are investigated.
Thesis (Ph.D.)--University of Adelaide, Dept. of Electrical and Electronic Engineering, 1995?
APA, Harvard, Vancouver, ISO, and other styles
You might also be interested in the bibliographies on the topic 'Speech Codes' for other source types:
Journal articles Books
We offer discounts on all premium plans for authors whose works are included in thematic literature selections. Contact us to get a unique promo code!

To the bibliography