Industrial design in endoscopy : the development of a tissue and organ extractor

Throughout history, developments in medicine have aimed to improve patient quality of life, and reduce the trauma associated with surgical treatment. Surgical access to internal organs and bodily structures has been traditionally via large incisions. Endoscopic surgery presents a technique for surgical access via small (1 Omm) incisions by utilising a scope and camera for visualisation of the operative site. Endoscopy presents enormous benefits for patients in terms of lower post operative discomfort, and reduced recovery and hospitalisation time. Since the first gall bladder extraction operation was performed in France in 1987, endoscopic surgery has been embraced by the international medical community. With the adoption of the new technique, new problems never previously encountered in open surgery, were revealed. One such problem is that the removal of large tissue specimens and organs is restricted by the small incision size. Instruments have been developed to address this problem however none of the devices provide a totally satisfactory solution. They have a number of critical weaknesses: -The size of the access incision has to be enlarged, thereby compromising the entire endoscopic approach to surgery. - The physical quality of the specimen extracted is very poor and is not suitable to conduct the necessary post operative pathological examinations. -The safety of both the patient and the physician is jeopardised. The problem of tissue and organ extraction at endoscopy is investigated and addressed. In addition to background information covering endoscopic surgery, this thesis describes the entire approach to the design problem, and the steps taken before arriving at the final solution. This thesis contributes to the body of knowledge associated with the development of endoscopic surgical instruments. A new product capable of extracting large tissue specimens and organs in endoscopy is the final outcome of the research.

Spectral coding methods for speech compression and speaker identification

This thesis investigates aspects of encoding the speech spectrum at low bit rates, with extensions to the effect of such coding on automatic speaker identification. Vector quantization (VQ) is a technique for jointly quantizing a block of samples at once, in order to reduce the bit rate of a coding system. The major drawback in using VQ is the complexity of the encoder. Recent research has indicated the potential applicability of the VQ method to speech when product code vector quantization (PCVQ) techniques are utilized. The focus of this research is the efficient representation, calculation and utilization of the speech model as stored in the PCVQ codebook. In this thesis, several VQ approaches are evaluated, and the efficacy of two training algorithms is compared experimentally. It is then shown that these productcode vector quantization algorithms may be augmented with lossless compression algorithms, thus yielding an improved overall compression rate. An approach using a statistical model for the vector codebook indices for subsequent lossless compression is introduced. This coupling of lossy compression and lossless compression enables further compression gain. It is demonstrated that this approach is able to reduce the bit rate requirement from the current 24 bits per 20 millisecond frame to below 20, using a standard spectral distortion metric for comparison. Several fast-search VQ methods for use in speech spectrum coding have been evaluated. The usefulness of fast-search algorithms is highly dependent upon the source characteristics and, although previous research has been undertaken for coding of images using VQ codebooks trained with the source samples directly, the product-code structured codebooks for speech spectrum quantization place new constraints on the search methodology. The second major focus of the research is an investigation of the effect of lowrate spectral compression methods on the task of automatic speaker identification. The motivation for this aspect of the research arose from a need to simultaneously preserve the speech quality and intelligibility and to provide for machine-based automatic speaker recognition using the compressed speech. This is important because there are several emerging applications of speaker identification where compressed speech is involved. Examples include mobile communications where the speech has been highly compressed, or where a database of speech material has been assembled and stored in compressed form. Although these two application areas have the same objective - that of maximizing the identification rate - the starting points are quite different. On the one hand, the speech material used for training the identification algorithm may or may not be available in compressed form. On the other hand, the new test material on which identification is to be based may only be available in compressed form. Using the spectral parameters which have been stored in compressed form, two main classes of speaker identification algorithm are examined. Some studies have been conducted in the past on bandwidth-limited speaker identification, but the use of short-term spectral compression deserves separate investigation. Combining the major aspects of the research, some important design guidelines for the construction of an identification model when based on the use of compressed speech are put forward.