Doppler ultrasound imaging modalities arguably represent one of the most complex task performed (usually in real time) by ultrasound scanners. At the heart of these techniques lies the ability to detect and estimate soft tissues or blood motion within the human body. As they have become an invaluable tool in a wide range of clinical applications, these techniques have fostered an intensive effort of research in the field of signal processing for more than thirty years, with a push towards more accurate velocity or displacement estimation. Coded excitation has recently received a growing interest in the medical ultrasound community. The use of these techniques, originally developed in the radar field, makes it possible to increase the depth of penetration in B-mode imaging, while complying with safety standards. These standards impose strict limits on the peak acoustic intensity which can be transmitted into the body. Similar solutions were proposed in the early developments of Doppler flow-meters to improve the resolution / sensitivity trade-off from which typical pulsed Doppler systems suffer. This work discusses the potential improvements in resolution, sensitivity and accuracy achievable in the context of modern Doppler ultrasound imaging modalities (taken in its broadest sense, that is, all the techniques involving the estimation of displacements, or velocities). A theoretical framework is provided for discussing this potential improvements, along with simulations for a more quantitative assessment. Colour Flow Imaging (CFI) modalities are taken as the main reference technique for discussion, due to their historical importance, and their relevance in many clinical applications. The potential achievable improvement in accuracy is studied in the context of modern velocity estimation strategies, which can be broadly classified into narrowband estimators (such as the “Kasai” estimator still widely used in CFI) and time shift based wideband strategies (normalised crosscorrelation estimator used, for instance, in applications like strain or strain rate estimation, elastography, etc.). Finally, simulations and theoretical results are compared to experimental data obtained with a simple custom-designed experimental set-up, using a single-element transducer.

[pt] Esta dissertação propõe algoritmos para codificações de voz a taxas médias em torno de 1,2 Kb/s. Um esquema de quantização vetorial preditiva chaveada com desempenho superior aos esquemas previamente descritos na literatura é proposto e avaliado em canal com ou sem ruído. Detectores eficientes de período fundamental e de sons oclusivos e fricativos são examinados e adaptados ao codificador proposto. Técnicas de exitação a baixas taxas de bits são investigadas a fim de reproduzir uma boa qualidade de voz decodificada. O modelo de exitação mista em multi-bandas com três sub-bandas é adotado para codificar os quadros sonoros. Para os quadros surdos são empregadas técnicas de modelagem e síntese de sinais fricativos e oclusivos, capazes de oferecer qualidade de voz satisfatória, reduzindo a taxa de bits destes quadros para apenas 0,4 Kb/s. Técnicas de pós-filtragem para reduzir o ruído de codificação e melhorar a qualidade de voz reconstruída são também examinadas e comparadas em uma mesma plataforma. Para reduzir o nível de ruído ambiente são ainda analisados métodos de supressão de ruído. Finalmente, o codificador proposto é comparado ao padrão norte-americano Mixed Excitation Linear Prediction (MELP), por meios de teste de comparação do tipo A/B. Os testes realizados indicam que o sistema proposto, operando a 1,2 Kb/s, apresenta qualidade de voz ligeiramente superior ao MELP, operando a 2,4 Kb/s. Para situações de transcodificação, o codificador proposto também apresenta desempenho superior ao MELP. / [en] This dissertation presents algorithms to encode at an avarage bit rate of 1.2 Kb/s. A novel switched-predictive vector quantiser technique that outperforms previously reported schemes is proposed and assessed under noise-free and noisy channels. Efficient detectors for the pitch period and fricative and stop sounds are examined and adapted to the proposed coder. Low bit rate excitation methods are investigated in order to reproduce rather high quality speech. A mixed multiband excitation approach with three sub-bands is employed to encode voiced frames. For unvoiced frames, fricatives and stops modelling and synthesis techniques are used. This approach has shown to provide high quality synthesised speech, whilts it reduces the bit rate to only 0.4 Kb/s for unvoiced frames. To reduce coding noise and improve decoded speech, post- filtering techniques are analysed and compared on the same plataform. To reduce background noise, noise suppression methods are also examined. Finally, the propose coder is evaluated against the North American Mixed Prediction (MELP) coder, through A/B comparison tests. Assessment results have shown that the proposed system, operating at 1.2 Kb/s, slightly outperformed the MELP coder, operating at 2.4 Kb/s. For tandem connection situations, the proposed algorithm has presented a superior performance than the MELP coder. / [es] Esta disertación propone algoritmos para codificaciones de voz a tasas medias en torno de 1,2 Kb/s. Se propone un esquema de cuantización vectorial predictiva, con desempeño superior a los esquemas previamente descritos en la literatura. Este esquema se evalúa en canal con o sin ruido. Se examinan detectores eficientes de período fundamental y de sueños oclusivos y fricativos se adaptan al codificador propuesto. Técnicas de exitación a bajas tasas de bits son investigadas a fin de reproducir una boa calidad de voz decodificada. Se adopta el modelo de exitación mixta en multi-bandas con tres sub-bandas para codificar los cuadros sonoros. Para los cuadros surdos se emplean técnicas de modelación y síntesis de señales fricativos y oclusivos, capaces de ofrecer calidad de voz satisfactoria, reduciendo la tasa de bits de estos cuadros para apenas 0,4 Kb/s. También se examinan y se comparan las técnicas de pós-filtragen para reducir el ruido de codificación y mejorar la calidad de voz reconstruída. Para reducir el nível de ruído ambiente se analizan métodos de supresión de ruido. Finalmente, el codificador propuesto se compara al padrón norteamericano Mixed Excitation Lineal Prediction (MELP), por medio de pruebas de comparación del tipo LA/B. Las pruebas realizadas indican que el sistema propuesto, operando a 1,2 Kb/s, presenta calidad de voz ligeramente superior al MELP, operando a 2,4 Kb/s. Para situaciones de transcodificación, el codificador propuesto también presenta desempeño superior al MELP.