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A novel transcoding algorithm for the adaptive multi rate (AMR) codec and the enhanced variable rate codec (EVRC) is proposed. In contrast to the conventional tandem transcoding algorithm, the proposed algorithm transcodes the parameters of one codec to the other without synthesizing the speech. The proposed algorithm decodes the parameters of source codec from the input bitstream, and based on frame classification and mode decision, it appropriately transforms the parameters of source codec to those of the target codec in the parametric domain. Finally, the transformed parameters are encoded into a bitstream that is decodable by the target codec. The parameters transcoded by the proposed algorithm are line-spectral pair (LSP), pitch delay, fixed codevector, codebook gains, and frame energy. Evaluation results show that while reducing both the computational complexity and delay by 50%, the proposed algorithm produces speech quality equivalent to that of produced by the tandem transcoding algorithm. The general idea is not restricted to the AMR and EVRC but is applicable to various other code-excited linear prediction (CELP) based codecs.