7 research outputs found

    New techniques in signal coding

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    Excitação multi-taxa usando quantização vetorial estruturada em árvore para o codificador CS-ACELP com aplicação em VoIP

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    Dissertação (mestrado) - Universidade Federal de Santa Catarina, Centro Tecnológico. Programa de Pós-Graduação em Engenharia Elétrica.Este trabalho apresenta um estudo sobre codificação multi-taxa estruturada sobre o algoritmo CS-ACELP (Conjugate-Structure Algebraic-Code-Excited Linear-Prediction) e a especificação G.729, cujo objetivo é propor um codificador com taxa variável, através da busca da melhor excitação fixa usando codebook estruturado em árvore, para aplicações VoIP (Voice-over-IP). A mudança progressiva do transporte de voz das redes de circuito para as redes IP (Internet Protocol), apesar dos diversos aspectos positivos, tem exposto algumas deficiências intrínsecas destas, mais apropriadas ao tráfego de #melhor esforço# do que ao tráfego com requisitos de tempo. Esta proposta está inserida no conjunto das iniciativas, no âmbito do transmissor, que procuram minimizar os efeitos danosos da rede sobre a qualidade da voz reconstruída. O codebook proposto tem estrutura em árvore binária, concebida a partir de uma heurística onde os vetores CS-ACELP são ordenados por valor de forma decrescente. Uma estratégia particular de armazenamento dos nós, envolvendo simplificação nos centróides, codificação diferencial e geração automática dos dois últimos níveis da árvore, permite reduzir o espaço de armazenamento de 640 para apenas 7 kwords. Através deste modelo chega-se a 13 taxas de codificação, de 5,6 a 8,0 kbit/s, com passo de 0,2 kbit/s. A relação sinal ruído fica em 1,5 dB abaixo da mesma medida na especificação G.729 para a taxa de 5,6 kbit/s, e apenas 0,6 dB abaixo quando na taxa 8,0 kbit/s. Testes subjetivos mostraram uma qualidade bastante aceitável para a taxa mínima e praticamente indistinguível do codec original na taxa máxima. Além disso, a busca da melhor excitação é 2,4 vezes mais rápida em comparação ao codec G.729 e pode ser totalmente compatível com este se a taxa for fixa em 8,0 kbit/s. This work presents a study about multi-rate coding structured over CS-ACELP (Conjugate-Structure Algebraic-Code-Excited Linear-Prediction) algorithm and G.729 standard, whose purpose is to come up with a variable rate codec by means of best fixed excitation search using a tree structured codebook, for VoIP (Voice-over-IP) applications. The progressive change of voice transmission from circuit switched to IP (Internet orks, besides its many positive aspects, has exposed some natural deficiencies of the latter, better suited to best effort traffics than traffics with time requirements. This proposition can be inserted in the bunch of efforts, related to the sender, that seek to reduce the network impairments over the quality of reconstructed voice. The suggested codebook has a binary tree structure heuristically conceived where algebraic CSACELP vectors are disposed by value in a decreasing order. Additionally, a particular approach to store the tree nodes are considered, which involves centroid implification, differential coding and automatic generation of the last two layers of the tree, squeezing the storing space from 640 down to 7 kwords. Through this model we reach 13 coding rates, ranging from 5.6 to 8.0 kbit/s, with 0.2 kbit/s step. The signal-to-noise ratio is 1.5 dB below the same measure for G.729 standard at the rate 5.6 kbit/s, and just 0.6 dB lower at 8.0 kbit/s. Subjective tests pointed to an acceptable quality at minimum rate and virtually indistinguishable quality from the original codec at the maximum one. Also, searching for the best fixed excitation is 2.4 times faster than G.729 and can be truly compatible with it if the rate is fixed in 8 kbit/s

    Proceedings of the Fifth International Mobile Satellite Conference 1997

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    Satellite-based mobile communications systems provide voice and data communications to users over a vast geographic area. The users may communicate via mobile or hand-held terminals, which may also provide access to terrestrial communications services. While previous International Mobile Satellite Conferences have concentrated on technical advances and the increasing worldwide commercial activities, this conference focuses on the next generation of mobile satellite services. The approximately 80 papers included here cover sessions in the following areas: networking and protocols; code division multiple access technologies; demand, economics and technology issues; current and planned systems; propagation; terminal technology; modulation and coding advances; spacecraft technology; advanced systems; and applications and experiments

    Évaluation subjective de la qualité (proposition d'un système de référence pour les codecs en bande élargie)

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    L'évolution des systèmes de télécommunications conduit à la conception de codecs de la parole et du son de plus en plus sophistiqués, accroissant ainsi la concurrence de l'industrie de l'audio et accordant une importance grandissante à la qualité de service. Si l'évaluation de la qualité des codecs peut s'opérer suivant des mesures objectives ou subjectives, les secondes restent les plus fiables dans la mesure où la qualité perçue par les utilisateurs est intrinsèquement subjective. Toutefois, les tests subjectifs requièrent des signaux d'ancrage, i.e. des signaux artificiels visant la reproduction des défauts perceptifs des codecs de sorte que les dégradations provoquées soient aisément contrôlables. Le système de référence actuellement normalisé par l'Union Internationale des Télécommunications est le MNRU (Modulated Noise Reference Unit) qui simule le bruit de quantification introduit par les premiers codecs en forme d'onde. L'évolution de la technologie rend aujourd'hui ce système obsolète, et il s'agit donc de concevoir un nouveau système d'ancrage plus adapté aux codecs actuels. En considérant la qualité audio comme un objet multidimensionnel, nous avons mis en évidence un espace perceptif à quatre dimensions, et ce à partir de deux approches de réduction de dimensionnalité, l'AFM (Analyse Factorielle Multiple) et la MDS 3 voies (MultiDimensional Scaling). A partir des quatre dimensions identifiées Réduction de la largeur de bande , Bruit de fond , Écho/Réverbération et Distorsion de la parole , nous avons modélisé puis validé les signaux d'ancrage des trois premières dimensions et proposé deux modèles de signaux d'ancrage pour la quatrième.The evolution of technology led to the design of very sophisticated speech and audio codecs. Accordingly, the competition in audio devices manufacturing has increased and today the quality of service becomes crucial for telecommunications operators. Quality of codecs is assessed through objective and subjective measures, the second ones being the most reliable since the quality perceived by users is inherently subjective. Nevertheless, subjective tests require anchor signals corresponding to artificial signals, which reproduce the perceptual impairments of codecs in such a manner that the amount of degradation can be easily controlled. The reference system currently standardized by the International Telecommunication Union is the Modulated Noise Reference Unit (MNRU), which simulates the quantization noise of the first generation of waveform codecs. Due to the evolution of codecs, the MNRU system became obsolete and researchers aim at designing a new reference system of anchor signals more suited to current codecs. Assuming that speech and audio quality is multidimensional, we first identified four perceptual dimensions using two dimensionality reduction techniques the MFA (Multiple Factor Analysis) and the 3 way MDS (MultiDimensional Scaling). From the identified dimensions, namely Bandwidth limitation , Background noise , Echo/Reverberation and Speech distortion , we succeeded in modeling and validating anchor signals for three of them and we suggested two models of anchor signals for the last one.RENNES1-Bibl. électronique (352382106) / SudocSudocFranceF

    HMM-based speech synthesis using an acoustic glottal source model

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    Parametric speech synthesis has received increased attention in recent years following the development of statistical HMM-based speech synthesis. However, the speech produced using this method still does not sound as natural as human speech and there is limited parametric flexibility to replicate voice quality aspects, such as breathiness. The hypothesis of this thesis is that speech naturalness and voice quality can be more accurately replicated by a HMM-based speech synthesiser using an acoustic glottal source model, the Liljencrants-Fant (LF) model, to represent the source component of speech instead of the traditional impulse train. Two different analysis-synthesis methods were developed during this thesis, in order to integrate the LF-model into a baseline HMM-based speech synthesiser, which is based on the popular HTS system and uses the STRAIGHT vocoder. The first method, which is called Glottal Post-Filtering (GPF), consists of passing a chosen LF-model signal through a glottal post-filter to obtain the source signal and then generating speech, by passing this source signal through the spectral envelope filter. The system which uses the GPF method (HTS-GPF system) is similar to the baseline system, but it uses a different source signal instead of the impulse train used by STRAIGHT. The second method, called Glottal Spectral Separation (GSS), generates speech by passing the LF-model signal through the vocal tract filter. The major advantage of the synthesiser which incorporates the GSS method, named HTS-LF, is that the acoustic properties of the LF-model parameters are automatically learnt by the HMMs. In this thesis, an initial perceptual experiment was conducted to compare the LFmodel to the impulse train. The results showed that the LF-model was significantly better, both in terms of speech naturalness and replication of two basic voice qualities (breathy and tense). In a second perceptual evaluation, the HTS-LF system was better than the baseline system, although the difference between the two had been expected to be more significant. A third experiment was conducted to evaluate the HTS-GPF system and an improved HTS-LF system, in terms of speech naturalness, voice similarity and intelligibility. The results showed that the HTS-GPF system performed similarly to the baseline. However, the HTS-LF system was significantly outperformed by the baseline. Finally, acoustic measurements were performed on the synthetic speech to investigate the speech distortion in the HTS-LF system. The results indicated that a problem in replicating the rapid variations of the vocal tract filter parameters at transitions between voiced and unvoiced sounds is the most significant cause of speech distortion. This problem encourages future work to further improve the system
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