自從MP3被發表以及風靡全世界之後,數位音訊技術變成日常生活中非常重要的ㄧ環。其相關的應用包含有數位音訊廣播、各類隨身聽、iPod、手機等等的產品。而MPEG組織接著MP3之後推出下ㄧ世代之音訊編碼技術MPEG-2/4 AAC，AAC比前ㄧ代的MP3有更高的壓縮率且能保持更好的音樂品質。不過在AAC編碼中採用了許多複雜的演算法及繁複的運算，而如何減低運算複雜度和維持相同音樂品質仍是一項挑戰，特別是在AAC編碼器端。
　　本論文針對AAC編碼器之關鍵模組，聲學模型做最佳化，聲學模型利用許多複雜的運算來模擬人類聽覺系統。本文採用memory-based 的架構來實現頻帶轉換模組，DSP導向架構來完成PAM，共享式記憶體及唯讀記憶體來降低硬體所需資源。在架構設計上，我們採用快速演算法及全管線式運算單元來提升系統效能，在低功耗考量上，我們採用快取式暫存器、gating-clock、Multi-Vth的設計來改善功耗問題。本文提出之架構設計在台積電0.13CMOS製程實現，需要43k邏輯閘，3.1MHz的操作頻率可達即時編碼動作，功耗約為3.67毫瓦。同時我們將我們的設計整合在SOC平台上並完成整體設計之驗證。

Since MP3 has been published, and became popular consumer applications, the digital audio technique is an important part in daily life. The applications of digital audio technique include broadcast system (DAB/DAB+), portable players, iPod, and mobile phone …etc. Organization of Moving Picture Experts Group (MPEG) proposed MPEG-2/4 AAC standard which is the audio encoding technique of next generation. Both the performance and compression ratio of AAC are better than MP3. However, the algorithm is more complex and computation-intensive. Hence, how to reduce the computation and maintain quality is a major challenge of AAC encoder.
In this thesis, we optimize the key component in MPEG-2/4 AAC encoder, which is psychoacoustic model (PAM). PAM has different complicated functions to model the human auditory system. This work exploits several methods to achieve low cost consideration, which are memory-based architecture for filterbank, DSP-oriented threshold generator, shared memory, and coefficient merged scheme. We use fully pipelined MDCT and fast algorithm for filterbank to improve performance. Moreover, we apply cache-register, clock-gating, operand isolation, and multi-Vth cell to save power consumption. As the synthesis result, our PAM consumes 43 k gate counts in TSMC 0.13 COMS technology, 3.1MHz operation frequency, 3.69mW for AAC encoder. Meanwhile, we also integrate our design into a SOC platform and perform the verification on the platform.

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