Digital signal processing in a 16kbps APC-AB codec by fixed point digital signal processor (FDSP-3)

Y. Tomita, S. Unagami, T. Taniguchi, Y. Tada, M. Taka
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引用次数: 0

Abstract

Recently much intensive research of 16kbps Speech coding algorithm has been conducted aiming to reduce the transmission bit rate and yet provides high speech quality. Adaptive predictive coding with adaptive bit allocation (APC-AB)[1] is considered to be one promising approach. However, the processing of this coding algorithm is so complicated that the implementation of the algorithm on a general-purpose signal processor, especially if fixed-point arithmetic DSPs are used, requires careful study of arithmetic operation precision and same way to reduce the number of processing cycles. Taking account of these points, real-time signal processing using a fixed-point signal processing chip (FDSP-3) has been studied, and a prototype codec has been realized. The prototype codec satisfied the CCITT mask of signal-to-total distortion ratio for PCM codecs and showed quality good enough for "toll" speech.
定点数字信号处理器(FDSP-3)在16kbps APC-AB编解码器中的数字信号处理
16kbps语音编码算法在降低传输比特率的同时又能保证较高的语音质量,近年来人们对16kbps语音编码算法进行了深入研究。带有自适应位分配的自适应预测编码(APC-AB)[1]被认为是一种很有前途的方法。然而,该编码算法的处理非常复杂,在通用信号处理器上实现该算法,特别是使用定点算术dsp时,需要仔细研究算术运算精度和减少处理周期的相同方法。考虑到这些问题,本文研究了采用定点信号处理芯片(FDSP-3)对信号进行实时处理,并实现了编解码器的原型。该原型编解码器满足PCM编解码器的信总失真比CCITT掩码,并显示出足够好的“收费”语音质量。
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