M. Noor, Kashif Israr Siddiqui, Muneeb Wahad Khan, Fraz Tajammul, Mazin Ashraf
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引用次数: 9
Abstract
During the history of digital audio compression there have been many different attempts at reducing the bit rate without reducing the equality in the same grade. Today we have many modem compression algorithms that achieve very high compression rations without a perceivable loss in quality. The main objective of our work is to implement a state of art speech-coding algorithm CS-ACELP using ITU-T's G.729 recommendation and optimize it for real-time implementation on a very long instruction word (VLIW) digital signal processor (DSP) central processing unit (CPU).