Spectral Envelope Modelling for Full-Band Speech Coding

Speech coding considering historically narrow-band was in the latest years significantly improved by widening the coded audio bandwidth. However, existing speech coders still employ a limited band source-filter model extended by parametric coding of the higher band. In this thesis, a full-band source-filter model is considered and especially its spectral magnitude envelope modelling. To match full-band operating mode, we modified, tuned and compared two methods, Linear Predictive Coding (LPC) and Distribution Quantization (DQ). LPC uses autoregressive modeling, while DQ quantifies the energy ratios between parts of the spectrum. Parameters of both methods were quantized with multi-stage vector quantization. Objective and subjective evaluations indicate the two methods used in a full-band source-filter coding scheme perform on the same range and are competitive against conventional speech coders requiring an extra bandwidth extension

Transmission efficace en temps réel de la voix sur réseaux ad hoc sans fil

La téléphonie mobile se démocratise et de nouveaux types de réseaux voient le jour, notamment les réseaux ad hoc. Sans focaliser exclusivement sur ces réseaux particuliers, le nombre de communications vocales effectuées chaque minute est en constante augmentation mais les réseaux sont encore souvent victimes d'erreurs de transmission. L'objectif de cette thèse porte sur l'utilisation de méthodes de codage en vue d'une transmission de la voix robuste face aux pertes de paquets, sur un réseau mobile et sans fil perturbé permettant le multichemin. La méthode envisagée prévoit l'utilisation d'un codage en descriptions multiples (MDC) appliqué à un flux de données issu d'un codec de parole bas débit, plus particulièrement l'AMR-WB (Adaptive Multi Rate - Wide Band). Parmi les paramètres encodés par l'AMR-WB, les coefficients de la prédiction linéaire sont calculés une fois par trame, contrairement aux autres paramètres qui sont calculés quatre fois. La problématique majeure réside dans la création adéquate de descriptions pour les paramètres de prédiction linéaire. La méthode retenue applique une quantification vectorielle conjuguée à quatre descriptions. Pour diminuer la complexité durant la recherche, le processus est épaulé d'un préclassificateur qui effectue une recherche localisée dans le dictionnaire complet selon la position d'un vecteur d'entrée. L'application du modèle de MDC à des signaux de parole montre que l'utilisation de quatre descriptions permet de meilleurs résultats lorsque le réseau est sujet à des pertes de paquets. Une optimisation de la communication entre le routage et le processus de création de descriptions mène à l'utilisation d'une méthode adaptative du codage en descriptions. Les travaux de cette thèse visaient la retranscription d'un signal de parole de qualité, avec une optimisation adéquate des ressources de stockage, de la complexité et des calculs. La méthode adaptative de MDC rencontre ces attentes et s'avère très robuste dans un contexte de perte de paquets

Transcoding between QCELP 13K and G.723.1 CELP speech coders

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

A Parametric Approach for Efficient Speech Storage, Flexible Synthesis and Voice Conversion

During the past decades, many areas of speech processing have benefited from the vast increases in the available memory sizes and processing power. For example, speech recognizers can be trained with enormous speech databases and high-quality speech synthesizers can generate new speech sentences by concatenating speech units retrieved from a large inventory of speech data. However, even in today's world of ever-increasing memory sizes and computational resources, there are still lots of embedded application scenarios for speech processing techniques where the memory capacities and the processor speeds are very limited. Thus, there is still a clear demand for solutions that can operate with limited resources, e.g., on low-end mobile devices. This thesis introduces a new segmental parametric speech codec referred to as the VLBR codec. The novel proprietary sinusoidal speech codec designed for efficient speech storage is capable of achieving relatively good speech quality at compression ratios beyond the ones offered by the standardized speech coding solutions, i.e., at bitrates of approximately 1 kbps and below. The efficiency of the proposed coding approach is based on model simplifications, mode-based segmental processing, and the method of adaptive downsampling and quantization. The coding efficiency is also further improved using a novel flexible multi-mode matrix quantizer structure and enhanced dynamic codebook reordering. The compression is also facilitated using a new perceptual irrelevancy removal method. The VLBR codec is also applied to text-to-speech synthesis. In particular, the codec is utilized for the compression of unit selection databases and for the parametric concatenation of speech units. It is also shown that the efficiency of the database compression can be further enhanced using speaker-specific retraining of the codec. Moreover, the computational load is significantly decreased using a new compression-motivated scheme for very fast and memory-efficient calculation of concatenation costs, based on techniques and implementations used in the VLBR codec. Finally, the VLBR codec and the related speech synthesis techniques are complemented with voice conversion methods that allow modifying the perceived speaker identity which in turn enables, e.g., cost-efficient creation of new text-to-speech voices. The VLBR-based voice conversion system combines compression with the popular Gaussian mixture model based conversion approach. Furthermore, a novel method is proposed for converting the prosodic aspects of speech. The performance of the VLBR-based voice conversion system is also enhanced using a new approach for mode selection and through explicit control of the degree of voicing. The solutions proposed in the thesis together form a complete system that can be utilized in different ways and configurations. The VLBR codec itself can be utilized, e.g., for efficient compression of audio books, and the speech synthesis related methods can be used for reducing the footprint and the computational load of concatenative text-to-speech synthesizers to levels required in some embedded applications. The VLBR-based voice conversion techniques can be used to complement the codec both in storage applications and in connection with speech synthesis. It is also possible to only utilize the voice conversion functionality, e.g., in games or other entertainment applications

Sparsity in Linear Predictive Coding of Speech

nrpages: 197status: publishe

Advanced signal processing techniques for pitch synchronous sinusoidal speech coders

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.EThOS - Electronic Theses Online ServiceGBUnited Kingdo

Um novo modelo para geração de tráfego VoIP

Resumo: Neste trabalho propõe-se um novo modelo para geração de tráfego VoIP. A inovação do modelo consiste em modelar o comportamento do usuário ao invés do tráfego agregado. O comportamento do usuário é retratado pela análise do tempo de retenção da chamada e pelo intervalo de tempo entre as mesmas. Enquanto a chamada está ativa, há geração de pacotes. O modelo da geração de pacotes consiste em caracterizar o tamanho dos pacotes e o intervalo de tempo entre os mesmos. As variáveis do modelo proposto foram caracterizadas por distribuições de probabilidades e modelos de séries temporais. Os dados utilizados para caracterizar o comportamento do usuário foram coletados no backbone de uma grande operadora de telecomunicações no Brasil, consistindo unicamente de chamadas VoIP. Para modelar a geração de pacotes, foram gerados dados em ambiente laboratorial. Este processo foi necessário para caracterizar unicamente a natureza dos dados. Um simulador de tráfego VoIP foi implementado e os resultados deste foram comparados com dados reais. A similaridade entre o tráfego real e sintético indica que o modelo funciona adequadamente e pode ser utilizado para estudos de redes VoIP e geração de carga. Adicionalmente foi desenvolvida uma aplicação que analisa a saída do simulador e torna mais eficiente o processo de modelamento de redes VoIP

The development of speech coding and the first standard coder for public mobile telephony

This thesis describes in its core chapter (Chapter 4) the original algorithmic and design features of the ??rst coder for public mobile telephony, the GSM full-rate speech coder, as standardized in 1988. It has never been described in so much detail as presented here. The coder is put in a historical perspective by two preceding chapters on the history of speech production models and the development of speech coding techniques until the mid 1980s, respectively. In the epilogue a brief review is given of later developments in speech coding. The introductory Chapter 1 starts with some preliminaries. It is de- ??ned what speech coding is and the reader is introduced to speech coding standards and the standardization institutes which set them. Then, the attributes of a speech coder playing a role in standardization are explained. Subsequently, several applications of speech coders - including mobile telephony - will be discussed and the state of the art in speech coding will be illustrated on the basis of some worldwide recognized standards. Chapter 2 starts with a summary of the features of speech signals and their source, the human speech organ. Then, historical models of speech production which form the basis of di??erent kinds of modern speech coders are discussed. Starting with a review of ancient mechanical models, we will arrive at the electrical source-??lter model of the 1930s. Subsequently, the acoustic-tube models as they arose in the 1950s and 1960s are discussed. Finally the 1970s are reviewed which brought the discrete-time ??lter model on the basis of linear prediction. In a unique way the logical sequencing of these models is exposed, and the links are discussed. Whereas the historical models are discussed in a narrative style, the acoustic tube models and the linear prediction tech nique as applied to speech, are subject to more mathematical analysis in order to create a sound basis for the treatise of Chapter 4. This trend continues in Chapter 3, whenever instrumental in completing that basis. In Chapter 3 the reader is taken by the hand on a guided tour through time during which successive speech coding methods pass in review. In an original way special attention is paid to the evolutionary aspect. Speci??cally, for each newly proposed method it is discussed what it added to the known techniques of the time. After presenting the relevant predecessors starting with Pulse Code Modulation (PCM) and the early vocoders of the 1930s, we will arrive at Residual-Excited Linear Predictive (RELP) coders, Analysis-by-Synthesis systems and Regular- Pulse Excitation in 1984. The latter forms the basis of the GSM full-rate coder. In Chapter 4, which constitutes the core of this thesis, explicit forms of Multi-Pulse Excited (MPE) and Regular-Pulse Excited (RPE) analysis-by-synthesis coding systems are developed. Starting from current pulse-amplitude computation methods in 1984, which included solving sets of equations (typically of order 10-16) two hundred times a second, several explicit-form designs are considered by which solving sets of equations in real time is avoided. Then, the design of a speci??c explicitform RPE coder and an associated eÆcient architecture are described. The explicit forms and the resulting architectural features have never been published in so much detail as presented here. Implementation of such a codec enabled real-time operation on a state-of-the-art singlechip digital signal processor of the time. This coder, at a bit rate of 13 kbit/s, has been selected as the Full-Rate GSM standard in 1988. Its performance is recapitulated. Chapter 5 is an epilogue brie y reviewing the major developments in speech coding technology after 1988. Many speech coding standards have been set, for mobile telephony as well as for other applications, since then. The chapter is concluded by an outlook

Survey of error concealment schemes for real-time audio transmission systems

This thesis presents an overview of the main strategies employed for error detection and error concealment in different real-time transmission systems for digital audio. The “Adaptive Differential Pulse-Code Modulation (ADPCM)”, the “Audio Processing Technology Apt-x100”, the “Extended Adaptive Multi-Rate Wideband (AMR-WB+)”, the “Advanced Audio Coding (AAC)”, the “MPEG-1 Audio Layer II (MP2)”, the “MPEG-1 Audio Layer III (MP3)” and finally the “Adaptive Transform Coder 3 (AC3)” are considered. As an example of error management, a simulation of the AMR-WB+ codec is included. The simulation allows an evaluation of the mechanisms included in the codec definition and enables also an evaluation of the different bit error sensitivities of the encoded audio payload.Ingeniería Técnica en Telemátic