Please use this identifier to cite or link to this item:
標題: 改善音訊切換品質與音訊播放增益之研究
A Study on the Improvements of Audio Switch Quality and Audio Replay Gain
作者: 白皓儒
Pai, Hao-Ju
關鍵字: 音訊切換;audio switch;音訊播放增益;短時間傅立葉轉換;Replay Gain;short time fourier transform
出版社: 電機工程學系所
引用: [1] David J. M. Robinson, “Perceptual model for assessment of coded audio”, A thesis submitted for the degree of Doctor of Philosophy Department of Electronic Systems Engineering University of Essex, March 2002 [2] D. Gabor, “Theory of communication”, J. Elect. Eng., vol. 93, pp.429-441, 1946 [3] Chen, V. C., H. Ling, “Time-Frequency Transforms for Radar Imaging and Signal Anslysis”, Boston: Artech House, 2002 [4] Sanjit K. Maitra, 貝蘇章教授審閱, 賴文能, 林國祥 ,高志暐譯著,”Digital Signal Processing: A Computer-Based Approach, 3e”, e高立圖書股份有限公司,2007年1月 [5] 王小川, 語音訊號處理, 全華圖書股份有限公司, 2009年2月 [6] 羅華強, 訊號處理-MATLAB的應用, 全華圖書股份有限公司, 2008年2月 [7] Leland B. Jackson, 嚴雨田譯著 , “Digital Filters and Signal Processing”, 茂昌圖書有限公司, 2002年7月 [8] Philip Denbigh, “System Analysis & Signal Processing”, Addison Wesley Longman Ltd,1998 [9] Samuel D. Stearns, “Digital Signal Processing with Examples in MATLAB”, CRC Press LLC, 2003 [10] WAI C. Chu, “SPEECH CODING ALGORITHMS”, John Wiley& Sons, Inc., 2003 [11] Jack Cartinhour, “DIGITAL SIGNAL PROCESSING : An Overview of Basic Principles”, Prentice-Hall, Inc., 2000 [12] Rai Reddy, “Spoken Language Processing-A Guide To Theory, Algorithm And System Development”, Prentice-Hall, Inc., 2001 [13] E. Zwicker & U.T. Zwicker, “Audio Engineering and Psychoacoustics Matching Signals to the Final Receiver, the Human Auditory System”, Journal of the Audio Engineering Society, Vol. 39, No. 3, pp. 115-126, 1991 [14] Robinson, D. W., Dadson, R. S., “A re-determination of the equal-loudness relations for pure tones.”, British Journal of Applied Physics, vol. 7, May, pp. 166-177., 1956 [15] Petre Stoica, Torsten Soderstrom, Optimal Instrumental Variable Estimates of the Parameters of an ARMA Process, IEEE TRANSACTION ON AUTOMATIC CONTROL,VOL., AC-30, N0.11, NOVEMBER,1985 [16] James A. Cadzow, “High Performance Spectral Estimation-A New ARMA Method”, IEEE TRANSACTIONS ON ACOUSTICS, SPEECH, AND SIGNAL PROCESSING, VOL., ASSP-28, N0.5, OCTOBER 1980 [17] Benjamin Friedlander, “Lattice Methods for Spectral Estimation”, PROCEEDING OF THE IEEE, VOK., 70, NO.9, SEPTEMBER 1982 [18] [19] [20] Sigang Qiu, “The Undersampled Discrete Gabor Transform”, IEEE TRANSACTION ON SIGNAL PROCESSING, VOL. 46, NO.5, MAY 1998
隨著多媒體產品價格的平民化,消費者多同時擁有數台數位影音產品。因此,發展出多輸入源的數位音訊設備。當這些音訊設備同時運作,在輸入源切換的過程中,會產生爆音的狀況。本文將針對此情形,做深入的探討與分析,並提出解決方式。除此之外,一般普通的音訊設備在播放不同音訊時,會衍生出響度不一樣的問題,透過調整[1]播放增益(Replay Gain),減少不同音訊間響度差異。
在第四章將介紹播放增益,使用播放增益調整音訊檔案,讓音訊響度達到一致。在第五章當中進行多個實驗以驗證第三章所提出的演算法,以及利用Borland C++ Builder軟體模擬播放增益效果。最後在第六章中針對各項實驗結果,提出改進的方式。

Because of the accessible price of multimedia products, consumers generally own more than two digital audio and video products. Subsequently, digital audio devices that have multiple inputs were developed. However, if these input functions are used simultaneously, these devices tend to emit a popping sound when switching between inputs. This study presents a discussion on the popping sound problem and proposes a solution to improve the audio switching quality. In addition, general audio devices tend to play different audio files at varying volumes. The proposed solution can reduce the volume variation for different audio files by using the Replay Gain standard.
Chapter 1 of this study introduces the multiple input switchers currently available on the market, which have the popping sound problem when switching between inputs. This chapter also presents a discussion on the playing of audio files at varying volumes. Chapter 2 introduces the time–frequency analysis methods employed in this study. This research investigated the cause and occurrence of the popping sound. The third chapter details the audio-processing technology used in this study. We developed an algorithm to improve the audio switching quality based on the analysis results obtained in Chapter 2.
Chapter 4 presents the Replay Gain standard. This study employed Replay Gain to adjust and equalize the audio volume. Chapter 5 details the experiments conducted to verify the algorithm introduced in Chapter 3. The Replay Gain results were simulated using Borland C++ Builder software. Finally, Chapter 6 provides suggestions to improve the experimental results.
Through this study, we hope to enhance the quality and development of audio processing methods.
其他識別: U0005-0108201221434100
Appears in Collections:電機工程學系所

Show full item record

Google ScholarTM


Items in DSpace are protected by copyright, with all rights reserved, unless otherwise indicated.