Low-Power Audio Input Enhancement for Portable Devices

With the development of VLSI and wireless communication technology, portable devices such as personal digital assistants (PDAs), pocket PCs, and mobile phones have gained a lot of popularity. Many such devices incorporate a speech recognition engine, enabling users to interact with the devices using...

Full description

Bibliographic Details
Main Author: Yoo, Heejong
Format: Others
Language:en_US
Published: Georgia Institute of Technology 2005
Subjects:
Online Access:http://hdl.handle.net/1853/6821
Description
Summary:With the development of VLSI and wireless communication technology, portable devices such as personal digital assistants (PDAs), pocket PCs, and mobile phones have gained a lot of popularity. Many such devices incorporate a speech recognition engine, enabling users to interact with the devices using voice-driven commands and text-to-speech synthesis. The power consumption of DSP microprocessors has been consistently decreasing by half about every 18 months, following Gene's law. The capacity of signal processing, however, is still significantly constrained by the limited power budget of these portable devices. In addition, analog-to-digital (A/D) converters can also limit the signal processing of portable devices. Many systems require very high-resolution and high-performance A/D converters, which often consume a large fraction of the limited power budget of portable devices. The proposed research develops a low-power audio signal enhancement system that combines programmable analog signal processing and traditional digital signal processing. By utilizing analog signal processing based on floating-gate transistor technology, the power consumption of the overall system as well as the complexity of the A/D converters can be reduced significantly. The system can be used as a front end of portable devices in which enhancement of audio signal quality plays a critical role in automatic speech recognition systems on portable devices. The proposed system performs background audio noise suppression in a continuous-time domain using analog computing elements and acoustic echo cancellation in a discrete-time domain using an FPGA.