| Summary: | 碩士 === 國立臺灣科技大學 === 電機工程系 === 92 === The new digital radio system DAB (Digital Audio Broadcasting), described in the European Eureka-147 standard, is a very innovative broadcast system and has attracted intensive attention in recent years because of its high quality audio service and supporting multimedia data to mobile, portable and fixed receivers. DAB will soon replace the existing analog broadcasting such as AM (Analog Modulation) and FM (Frequency Modulation) in the future.
Audio compression is a requirement for efficient transmission and storage of high fidelity digital audio. In this thesis, we focus on the DAB audio decoding (ISO/IEC MPEG Audio Layer II Decoding) and record the audio data for further listening. The functions and computation complexity in DAB audio decoder will be analyzed to reduce and simplify the decoding overhead. The proposed architecture has been integrated into one hardware block and successfully coded in synthesizable Verilog HDL language and verified through FPGA environment. Further, we look at other audio compression standards with regards to both compression rate and sound quality. The results obtained with different algorithms are discussed and compared. The chosen recording format, MPEG/Audio Layer III (MP3), is improved on its encoding complexity. A software program of the DAB audio recorder based on a modified real-time MP3 encoder is built.
The proposed design of the audio decoder and recorder of a DAB receiver retains both the decoding computation capability in the hardware and the recording flexibility in the software. The achievement of this thesis can be used as a basis of future applications and speed up the popularity of DAB receivers.
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