目前最常用的音頻訊號壓縮處理技術包括MP3、CELP、FLAC等，這些技術將多個取樣的訊號點以區塊為基礎(block-based)作分析壓縮編碼處理，導致顯著的時間延遲(端點至端點延遲通常大於20 msec)，並不適合某些特殊應用情境，例如錄音室或實況表演使用的無線麥克風系統，其延遲要求低於5 msec，因此本論文選用具低延遲特性的ADPCM技術作為音頻壓縮編碼處理。目前國際電信聯盟(ITU)制訂的ADPCM標準G.726只針對語音訊號(訊號頻寬小於4KHz，每個取樣點為8bits)，所以本論文參考此標準提出可對音頻訊號(訊號頻寬約20KHz)做壓縮編碼的演算法。針對無線數位麥克風的標準，將每個取樣點24 bits (取樣頻率為48KHz)的輸入訊號，經過本論文設計的適應性量化(adaptive quantization)與適應性預測(adaptive prediction)區塊處理壓縮成16 bits或12 bits後傳送出去，在接收端重建回來後依然保有良好的訊號品質。

In this thesis the adaptive differential pulse code modulation (ADPCM) with low latency is adopted for the audio compression coding. This thesis extends the G.726 standard and proposes a novel compression algorithm for the audio signals which bandwidth is about 20 KHz. For the typical wireless digital microphone system, the audio signal is converted to a digital data with 48K samples per second and 24 bits per sample. This input digital signal can be compressed to 16 bits or 12 bits per sample through the proposed process which contains adaptive quantization and adaptive prediction. Simulation results show that the reconstructed signal at the receiver still has a good quality.

[1] ITU Recommendation G.726, “40, 32, 24, 16 kbit/s Adaptive Differential Pulse Code Modulation (ADPCM),” 1990.
[2] Madeleine Bonnet, Odile Macchi and M. Jaidane-Saidane, “Theoretical analysis of the ADPCM CCITT Algorithm,” IEEE Trans. On Communications, vol. 38, n6, pp.847-858, June 1990.
[3] C.R. Johnson, “Adaptive IIR Filter(s) In the CCITT 32 KBPS ADPCM Standard,” IEEE Conference on Circuits, Systems and Computers, Nov. 1985.
[4] Simon Haykin, Communication Systems, 4th ed., New York: Wiley, 2001.
[5] Zoran H. Peric, Jelena R. Nikolic and Aleksandra Z. Jovanovic, “Design of piecewise uniform scalar quantizer for the backward adaptive algorithm,” IEEE International Conference on Telecommunication in Modern Satellite, Cable, and Broadcasting Services, Oct. 2009.
[6] P. L. Feintuch, “An adaptive recursive LMS filter,” Proc. IEEE, vol. 64, pp. 1622-1624, NOV. 1976.
[7] Simon Haykin, Adaptive Filter Theory, 5th ed., Prentice-Hall, 2014.
[8] J. Dhiman, S. Ahmad and K. Gulia, “Comparison between Adaptive filter Algorithms (LMS,NLMS,RLS),” International Journal of Science, Engineering and Technology Research (IJSETR), Volume 2, Issue 5,2013.
[9] Simon Haykin, Bernard Widrow (Editor), Least-Mean-Square Adaptive Filters, Wiley, 2003.
[10] B. Koo and J. D. Gibson, “Experimental comparison of all-pole, all-zero, and pole-zero predictors for ADPCM speech coding,” IEEE Trans. On Communications, vol. COM-34, pp. 285-290, Mar. 1986.
[11] N.S. Jayant and P. Noll, Digital Coding of Waveforms, Principles and Applications to Speech and Video, Prentice Hall Secondary Education Division, New Jersey, 1984.