Abstract
dc:descriptionThe approach presented in these pages processes the audio information digitally and generates a two-level pulse-width modulation (PWM) signal that is amplified by a switching power stage and finally low-pass filtered before it is sent to a speaker. The purpose of the digital signal processing (DSP) portion of the system is to transform the original audio information into a feasible, low-level PWM signal, with minimum possible degradation. The digital audio information is first upsampled, then converted from uniform to natural sampling to minimize distortion, and finally noise shaped to reduce the time resolution needed by the PWM edges without compromising the signal-to-noise (SNR) ratio. A PWM generator converts samples into pulses, and a switching stage creates a high-power replica of the PWM signal. Switching minimizes energy losses and introduces minimum extra degradation. The output low-pass filter extracts the audio information from the PWM signal and delivers it to the speaker. Each of the functional blocks required by this approach has been justified, implemented, and evaluated.
Degree
thesis:*- Name thesis:degree_name
- Ph.D.
- Level thesis:degree_level
- Dissertation
- Discipline thesis:degree_discipline
- Electrical Engineering
- Grantor
- University of Illinois at Urbana-Champaign
- Year dc:date
- 2015
Author and committee
dc:creator, dc:contributor.*- Author dc:creator
-
- Pascual, Cesar
- Contributors dc:contributor
-
- Krein, Philip T.
Subjects
dc:subject × 1Rights
- Language dc:language
- eng
Identifiers
dc:identifier.*- Identifier
- (MiAaPQ)AAI9996671
- OAI identifier oai:identifier
- oai:www.ideals.illinois.edu:2142/81371