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姓名 楊東敏(Dong-Min Yang) 查詢紙本館藏 畢業系所 通訊工程學系 論文名稱 基於線性預測編碼及音框基頻週期同步之高品質語音變換技術
(High Quality Speech Transformation Based on Linear Prediction Coding and Pitch Synchronization)相關論文 檔案 [Endnote RIS 格式]
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摘要(中) 語音變換(speech transformation)包含可以改變語音的音調以及說話的速度，語音的資料可利用語音變換技術，讓平凡單調的聲音變得豐富有趣，也可以做為保密或隱私之應用，使得語音處理的應用變得更多元化。
最理想的語音變換方法是使用語音分析合成(analysis-synthesis)的方式，獲得語音基本的重要特徵，藉由這些特徵可以很彈性的合成出高品質的語音；然而，當基頻週期或音框長度被改變時，以一個固定音框(frame)來進行語音分析合成，會造成基頻週期不同步的嚴重問題。為此，本論文提出利用線性預測編碼(linear prediction coding)的分析合成技術結合音框基頻週期同步(pitch synchronization)的機制，來解決此問題；針對有聲音框較穩定的特性，依LPC殘餘訊號(residual)及口腔頻譜(formant spectral)係數的特徵，調整基頻週期及口腔頻譜係數，或者改變音框訊號的長度，結合以交互相關法(cross-correlation)的音框基頻週期同步機制，來合成變換後語音的輸出訊號。經由模擬實驗的結果證實，語音音調特性的改變，及合成的語音品質，皆令人滿意，尤其是女聲變男聲之效果更佳。摘要(英) Speech transformation that can change the tone and the speed of a speaker is useful in privacy protection and entertainment. Intuitively, the best approach for speech transformation is the analysis-synthesis method because it has the freedom for synthesizing arbitrary voice once the speech parameters are obtained. However, pitch out of synchronization is a serious problem in most frame-based speech analysis and synthesis processes when the pitch or the frame length is changed.
In this paper we propose a speech transformation method based on the linear prediction coding with pitch synchronization to solve the problem. The speech transformation is only performed in voiced frames because they are relatively stable and have high energy. Cross-correlation is calculated to locate the synchronization point, i.e., the pitch mark. Arbitrary pitch scaling and time scaling can then be performed. Simulations show that the speech can be modified to different timbre and tone, or speaking speed, with high quality, especially in the case of transforming from female to male.關鍵字(中) ★ 語音變換
★ 線性預測編碼
★ 同步
★ 基頻週期關鍵字(英) ★ pitch
★ linear prediction coding
★ synchronization
★ speech transformation論文目次 第一章緒論………………………………………….………… 1
1.1 研究動機……….………………..………………………….. 1
1.2 論文架構……………….……………….…..………………... 2
第二章語音變換技術的發展…………………….……… 5
2.1 語音變換的發展與應用………...…….……………...….… 5
2.2 語音變換技術的基本概念….……………………….…… 6
2.3 本語音變換技術的系統架構……………………….…. 8
第三章語音變換技術的簡介………………………....… 15
3.1 語音的特性…………………………...………..…………... 15
3.2 語音變換技術……………………………………………… 17
3.2.1 時間變速技術……………….....…….……………..…… 18
3.2.2 基頻週期變調技術………….....…….……………..…… 19
3.3 線性預測編碼…………………….……………………….. 24
3.4 開迴路基頻分析………………….……………………….. 30
3.5 語音品質評估方法……………….………………………..33
第四章基於線性預測編碼之語音變換技術……..…. 35
4.1 有聲/無聲的決策…………..…………………....…...……. 36
4.2 線性預測編碼的分析與調整……...……………….……..39
4.3 基頻週期的分析…………………....……………….……..43
4.3.1 基頻週期的估計…….………………..……………..…… 43
4.3.2 基頻週期標記的搜尋…..……………..……………..…… 46
4.4 音框基頻週期同步機制…………....……………….……..49
4.4.1 截取分析音框長度…….……………..……………..…… 49
4.4.2 搜尋基頻週期同步位置.……………..……………..…… 51
4.5 音框臨界補償………………….…....……………….……..56
4.6 語音變調/基頻週期調整……………….…....…………….60
4.6.1 基頻週期壓縮.……………..…………………………… 61
4.6.2 基頻週期延伸.……………..…………………………… 64
4.7 語音變速/說話速度調整……………….…....…………….67
4.7.1 說話時間縮短/速度加快…..…………………………… 67
4.7.2 說話時間延長/速度減慢…..…………………………… 69
第五章模擬實驗與品質評估………….……………….. 74
5.1 模擬環境及語音資料…..……………………………...… 74
5.2 有聲/無聲的決策分析……………...………………..…..… 74
5.3 不同口腔頻譜的調整……...……………...…..………...… 78
5.4 語音變換的品質評估……………………………….… 80
5.4.1 基頻週期縮短的品質評估…..…………………………… 80
5.4.2 基頻週期延伸的品質評估…..…………………………… 83
5.4.3 說話速度加快的品質評估…..…………………………… 85
5.4.4 說話速度減慢的品質評估…..…………………………… 87
第六章結論……………………………………….….……..… 89
參考文獻………………………………………………………..… 90參考文獻 [1]M. Edgington, A. Lowry, "Residual-Based Speech Modification Algorithms for Text-to-Speech Synthesis, " BT Labaratories, Martlesham Heath, IPSWICH, IP5 7RE, U.K.
[2]ITU-T Recommendation G.729, "Coding of speech at 8 kbit/s using Conjugate-Structure Algebraic-Code-Excited Linear-Prediction (CS-ACELP)," March 1996.
[3]B. Gold, N. Morgan, Speech and Audio Signal Processing, Wiley, 2000
[4]A. M. Kondoz, Digital Speech Coding for Low Bit Rate Communications Systems, Wiley, 1994.
[5]J. Makhoul, "Linear Prediction: a tutorial review," Proceedings of the IEEE, vol. 63, 1975, pp. 561-580.
[6]ITU-T Recommendation on P.800, "Methods for subjective determination of transmission quality," Aug. 1996.
[7]王鴻彬, 國語聲訊處理, 碩士論文, 國立交通大學, 1995
[8]J. Laroche, "Autocorrelation Method for High-Quality Time/Pitch-Scaling," 1993 IEEE Workshop on Applications of Signal Processing to Audio and Acoustics, Oct. 1993.
[9]G. J. Lin, S. G. Chen, T. Wu, "High Quality and Low Complexity Pitch Modification of Acoustic Signals," ICASSP, vol. 5, pp. 2987-2990, 1995.
[10]J. Slifka, T. R. Anderson, "Speaker Modification with LPC Pole Analysis," ICASSP-95, 1995 International Conference on Acoustics, Speech, and Signal Processing, vol. 1, pp. 644 –647, May 1995.
[11]F. Thomas, J. Robert, "Shape Invariant Time-Scale and Pitch Modification of Speech," IEEE Transactions on Signal Processing, vol. 40, No. 3, March 1992.
[12]P. Rodriguez-Peralta, M. Nakano-Miyatake, H. Perez-Meana,G. Duchen-Sanchez, "Time Scaling Algorithm of Speech Signal to Assist Learning of a Foreign Language," ISIE 2000. Proceedings of the 2000 IEEE International Symposium on Industrial Electronics, vol. 2, pp. 586-589, Dec. 2000.
[13]翁正平, 具語者轉換功能之語音編碼器, 碩士論文, 國立台北科技大學, 1999
[14]B. Atal, L. Rabiner, "A Pattern Recognition Approach to Voiced-Unvoiced-Silence Classification with Applications to Speech Recognition," IEEE Transactions on Acoustics, Speech, and Signal Processing, vol. 24, pp. 201 –212, Jun 1976.指導教授 張寶基(Pao-Chi Chang) 審核日期 2003-7-10 推文 plurk
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