高品質切換式離散餘弦與小波封包
轉換之音訊編碼技術

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姓名

何恭岳(Kong-Yueh Ho) 查詢紙本館藏

畢業系所

通訊工程學系

論文名稱

高品質切換式離散餘弦與小波封包轉換之音訊編碼技術
(High Quality Switched Discrete CosineTransform and Wavelet PacketAudio Coding Technique )

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摘要(中)

轉換編碼應用於音訊處理系統中已行之有年，而近年來最熱門的分頻編碼法則莫過於小波轉換。其多層解析之特性，使得分頻處理之選擇架構更為多元。本論文提出一套混合式高音質壓縮系統，在處理每個音框資料時，首先即依據頻率域之平坦度量測以決定使用頻率解析度較佳之離散餘弦轉換或具豐富時間資訊之小波轉換作為此音框的主要轉換方式。對於採用離散餘弦轉換之音框，將依循人耳聲學模型所算出之頻域遮罩配合非線性量化器作量化。若是選擇運用小波封包做分頻，則樂音訊號通過小波濾波器組後即分成26個固定子頻帶，並在各子頻帶之後再依據小波域與頻率域平坦度之量測選擇性地使用離散餘弦轉換以提升頻率解析度。配合針對非理想濾波器組之最佳化位元分配演算法，將頻域上之人耳聽覺遮蔽曲線轉換為小波域之遮蔽曲線，以提供精良之量化準則並保有極高之音質。最後再以熵編碼，將量化後係數封裝成位元流。實驗結果顯示，在64k位元率的情況下，本系統所提供之音質，不僅優於MP3，更能超越AAC低複雜度規格。

摘要(英)

We propose a hybrid coding system that utilizes both Wavelet Packet (WP) and DCT techniques. To process each audio frame, the system selects either WP or DCT to process based on the frame flatness measures in wavelet domain and frequency domain. If DCT is chosen, all DCT coefficients are quantized by a non-uniform quantizer according to the frequency masking curve. On the other hand, frame data are segmented into 26 fixed subbands when WP is chosen. Then, the system selectively utilizes DCT to promote frequency resolution of each subband based on the subband flatness measure. By quoting optimal bit-allocation for non-ideal filter bank, the masking threshold from psychoacoustic model can be translated into specific criteria in the wavelet domain for quantization. Experiment results show that the proposed system is superior to MP3 and AAC LC profile at 64k bps

關鍵字(中)

★ 最佳化位元分配
★ 離散餘弦轉換
★ 小波濾波器

關鍵字(英)

★ optimal bit-allocation
★ DCT
★ Wavelet Packet

論文目次

目錄 IV
圖目 VII
表目 X
第一章緒論 1
1.1音訊壓縮簡介 1
1.2研究動機與目的 3
1.3系統架構 5
1.4論文架構 7
第二章小波分析技術 8
2.1小波轉換（Wavelet Transform） 8
2.1.1小波分解（Wavelet Expansion）與離散小波轉換（Discrete Wavelet Transform） 9
2.1.2多重解析度分析 10
2.2小波轉換與數位訊號處理 12
2.2.1 小波濾波器 12
2.2.2 小波轉換 18
2.2.3 小波封包（Wavelet Packet） 19
2.3 算數編碼（Arithmetic coding） 21
第三章人類聽覺心理學模型及其應用實例 24
3.1現代音訊編碼技術 24
3.2人耳聲學模型基本原理與其應用 25
3.2.1人耳最小感知音壓與臨界頻帶(Critical Band) 26
3.2.2頻率軸上的遮蔽效應 29
3.2.3時間軸上的遮蔽效應 36
3.2.4模型公式 37
3.3 MPEG – 2 AAC之簡介 40
3.3.1濾波器組（Filter Bank） 41
3.3.2時域雜訊塑型（Temporal Noise Shaping; TNS） 42
3.3.3預測（Prediction） 43
3.3.4 M/S立體聲與強度編碼（M/S Stereo and Intensity Encoding） 43
3.3.5量尺因子頻帶Scale Factor Bands 44
3.3.6量化（Quantization） 44
3.3.7無失真編碼（Noiseless Coding） 45
3.3.8位元流的格式（Bit-stream Format） 45
第四章混合式音訊壓縮系統 47
4.1 時頻轉換（Time/Freq. Mapping） 48
4.1.1 小波封包轉換 49
4.1.2 離散餘弦轉換 50
4.2 平坦度量測（Flatness Measure） 51
4.3 最佳化位元配置 54
4.4 量化 58
第五章實驗結果與討論 59
5.1客觀評量工具-EAQUAL簡介 59
5.2測試歌曲說明 61
5.3小波封包與ISO – Standard MP3之比較 63
5.4混合小波封包轉換與離散餘弦轉換 64
5.5混合式轉換與現代音訊編碼器之比較 68
第六章結論與未來展望 71
6.1結論 71
6.2未來展望 71
參考文獻 73

參考文獻

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指導教授

張寶基(Pao-Chi Chang)

審核日期

2005-6-24

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