| Summary: | 碩士 === 國立成功大學 === 電機工程學系碩博士班 === 91 === The MPEG audio compression is a lossy audio compression method, which provides high quality audio under high compression ratio. MPEG 1 audio standard provides three layers, layer I, layer II, and layer III. Layer I is the simplest but low audio quality, while layer III have the most complexity but offers the best audio quality. The MPEG audio standard offers four channel modes, which can be one of mono mode, dual channel mode, stereo mode, or joint stereo mode. The sampling frequency could be 32kHz, 44.1kHz, or 48kHz. Because of the high compression ratio and high sound quality, the MPEG1 layer III(MP3) has been used by network users as a sound compression method in many multimedia applications nowadays.
In this thesis, we design a MP3 audio system. From experiment results of the complexity analysis, MP3 encoder/decoder system could be divided into two parts, high control complexity part (such as Huffman coding, quantization/invert quantization, and stereo processing) and high computation complexity part (such as poly phase analysis/synthesis filter bank, MDCT/IMDCT). We implement the high control complexity part by software, and use hardware to implement the high computation complexity part. Finally, we apply software/hardware co-simulation to verify the entire MP3 codec system. We choose ARM integrator to implement and verify the MP3 system. ARM integrator is a complete environment for SoC design, which uses ARM bus as the transform protocol. In the software implementation, it requires 26 minutes and 43 seconds to compress an audio source(wave format) with 630kB to a MP3 format. However, it only requires 1 minute and 48 seconds after hardware/software co-design. The experiment results show it improves the system performance.
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