M. Tech. (Chemical Engineering, Faculty of Engineering and Technology), Vaal University of Technology / Amplitude modulation is not commonly associated with effective amplifying. This work focuses on implementing amplitude modulation into a high efficiency Class E amplifier. Different types of amplifiers are compared with each other, to show the advantages of using a Class E amplifier. The theory of the Class E amplifier is dealt with in detail. A harmonic filter is designed for the amplifier to make it radio spectrum friendly. The modulation process is implemented with the aid of a transformer into the Class E amplifier. The advantage of this is that the transformer serves both as a radio frequency choke for the Class E circuit as well as a modulator. The implementation of the amplitude modulation into the high efficient Class E circuit was successful. The final Class E circuit had superb efficiency, the harmonic filter showed good harmonic attenuation and the modulation process had low distortion. All this resulted in a fine low power AM transmitter.

M.Ed. (Media Studies) / Please refer to full text to view abstract

This study examined the changes that occur in spontaneous speech when speakers are distracted by background audio. Forty young adults answered open ended questions under five audio conditions (pink noise, movie dialogue, heated debate, classical music, and contemporary music) and a silent condition. Acoustic parameters assessed during the sessions included mean and variability of the fundamental frequency (F0), mean and variability of intensity, speaking time ratio, and disfluency ratio. It was hypothesized that there would be significant increases in the mean and variability of F0 as well as the mean and variability of intensity. There were statistically significant increases in mean and variability of intensity and mean F0 across most audio conditions. There were no significant changes in variability of intensity in the pink noise condition and no significant changes in variability of F0 in any audio condition. We hypothesized that the speaking time ratio would decrease in the presence of background audio compared to the silent baseline. Results demonstrate significant increases in speaking time ratio except for the classical music condition. It was expected that the disfluency ratio of speech production for each participant would increase in the presence of background audio, with informational masking demonstrating the most increase. Results revealed a significant increase in disfluency ratios across background audio conditions except for the pink noise and classical music conditions. Participants reported the heated debate and contemporary music to be the most distracting. These results have potential clinical implications regarding the type of environment where therapy is given, and what type of everyday situations might cause the most difficulties with fluency as well as the processing and production of speech.

No description available.

En el presente trabajo de tesis se presenta el diseño, implementación y los resultados de las pruebas de un sistema de amplificación, filtrado en frecuencia y adquisición de datos para un arreglo de tres hidrófonos de tipo piezoeléctrico que serán utilizados por un vehículo submarino operado remotamente. El sistema debe monitorear hasta tres hidrófonos piezoeléctricos teniendo control de ganancia para los tres canales, siendo la máxima de por lo menos 40dB. El ancho de banda de los amplificadores abarca desde 1 Hz hasta 100KHz. Cada canal cuenta con cuatro filtros de 4to orden; dos pasa-altos de 0.1 y 10 Hz y dos pasa-bajos de 100 Hz y 30 KHz. Los cuáles serán seleccionables en base a los requerimientos de las pruebas a realizar. Un sistema de digitalización de señales permite capturar las ondas dentro de su ancho de banda. Todo esto se encuentra energizado por un sistema de regulación de voltaje de bajo ruido para asegurar la pureza de la señal capturada. Dado que este sistema fue desarrollado para formar parte de un vehículo submarino operado remotamente (ROV), su diseño e implementación también responden a la necesidad de acoplarse en el recipiente de forma cilíndrica en el cual va a ser alojado. Este arreglo además de capturar la información acústica para luego ser almacenada y procesada, permitirá determinar la localización relativa de la fuente de sonido mediante la triangulación de las señales obtenidas por el arreglo de hidrófonos, del cual se presentan resultados experimentales que demuestran la viabilidad del concepto. / Tesis

This thesis presents the theory, implementation and applications of the harmonic sinusoid modeling of pitched audio events. Harmonic sinusoid modeling is a parametric model that expresses an audio signal, or part of an audio signal, as the linear combination of concurrent slow-varying sinusoids, grouped together under harmonic frequency constraints. The harmonic sinusoid modeling is an extension of the sinusoid modeling, with the additional frequency constraints so that it is capable to directly model tonal sounds. This enables applications such as object-oriented audio manipulations, polyphonic transcription, instrument/singer recognition with background music, etc. The modeling system consists of an analyzer and a synthesizer. The analyzer extracts harmonic sinusoidal parameters from an audio waveform, while the synthesizer rebuilds an audio waveform from these parameters. Parameter estimation is based on a detecting-grouping-tracking framework. The detecting stage finds and estimates sinusoid atoms; the grouping stage collects concurrent atoms into harmonic groups; the tracking stage collects the atom groups at different time to form continuous harmonic sinusoid tracks. Compared to standard sinusoid model, the harmonic model focuses on harmonic groups of atoms rather than on isolated atoms, therefore naturally represents tonal sounds. The synthesizer rebuilds the audio signal by interpolating measured parameters along the found tracks. We propose the first application of the harmonic sinusoid model in digital audio editors. For audio editing, with the tonal events directly represented by a parametric model, we can implement standard audio editing functionalities on tonal events embedded in an audio signal, or invent new sound effects based on the model parameters themselves. Possibilities for other applications are suggested at the end of this thesis.

This thesis presents research into reducing microphone artefacts in live sound with no prior knowledge of the sources or microphones. Microphone artefacts are defined as additional sounds or distortions that occur on a microphone signal that are often undesired. We focus on the proximity effect, comb filtering and microphone bleed. In each case we present a method that either automatically implements human sound engineering techniques or we present a novel method that makes use of audio signal processing techniques that goes beyond the skills of a sound engineer. By doing this we can show that a higher quality mix of a live performance can be achieved. Firstly we investigate the proximity effect which occurs on directional microphones. We present a method for detecting the proximity effect with no prior knowledge of the source to microphone distance. This then leads to a method for reducing the proximity effect which employs a dynamic filter informed by audio analysis. Comb filtering occurs when the output of microphones reproducing the same source are mixed together. We present a novel analysis of how the accuracy of a technique to automatically estimate the correct delay of the source between each microphone is affected by source bandwidth and the windowing function applied to the data. We then present a method for reducing microphone bleed in the multiple source, multiple microphone case, both in determined and overdetermined configurations. The proposed method is extended from prior research in noise cancellation, which has not previously been applied to musical sound sources. We then present a method for simulating microphone bleed in synthesised drum recordings, where bleed enhances the realism of the output. Through subjective listening tests and objective measures each proposed method is shown to succeed at reducing the microphone artefacts while preserving the original sound source.

Digitized by Kansas Correctional Industries

Digitized by Kansas Correctional Industries

Digitized by Kansas Correctional Industries