A novel transcoding algorithm for SMV and g.723.1 speech coders via direct parameter transformation
نویسندگان
چکیده
In this paper, a novel transcoding algorithm for the Selectable Mode Vocoder (SMV) and the G.723.1 speech coder is proposed. In contrast to the conventional tandem transcoding algorithm, the proposed algorithm converts the parameters of one coder to the other without going through the decoding and encoding process. The proposed algorithm is composed of four parts: the parameter decoding, Line Spectral Pair (LSP) conversion, pitch period conversion and rate selection. The evaluation results show that the proposed algorithm achieves equivalent speech quality to that of tandem transcoding with reduced computational complexity and delay.
منابع مشابه
An efficient transcoding algorithm for g.723.1 and g.729a speech coders
To set a valid communication channel between two endpoints employing different speech coders, decoder and encoder of each endpoint need to be placed in tandem. However, tandem coding is often associated with problems such as poor speech quality, high computational load, and additional transmission delay. In this paper, we propose an efficient transcoding algorithm for a legitimate communication...
متن کاملA novel rate selection algorithm for transcoding CELP-type codec and SMV
In this paper, we propose an efficient rate selection algorithm that can be used to transcode speech encoded by any code excited linear prediction (CELP)-type codec into a format compatible with selectable mode vocoder (SMV) via direct parameter transformation. The proposed algorithm performs rate selection using the CELP parameters. Simulation results show that while maintaining similar overal...
متن کاملTranscoding algorithm for g.723.1 and AMR speech coders: for interoperability between voIP and mobile networks
In this paper, an efficient transcoding algorithm between G.723.1 and AMR speech coders is proposed for providing interoperability between IP and mobile networks. Transcoding is completed through three processing steps: line spectral pair (LSP) conversion, pitch interval conversion, and fast adaptivecodebook search. For maintaining minimum distortion, sensitive parameters to quality such as ada...
متن کاملA voicing-driven packet loss recovery algorithm for analysis-by-synthesis predictive speech coders over Internet
In this paper, a novel voicing-driven adaptive packet loss recovery algorithm is proposed to lessen the possible voice degradation and error propagation for analysis-by-synthesis speech coders in Internet applications. After voicing classification, we adaptively adopt random noise generation, multiresolution excitation generation, or pulse tracking procedure to recover the lost packets. By appl...
متن کاملAn Efficient Transcoding Scheme for G.729 and G.723.1 Speech Codecs: Interoperability over the Internet
This paper proposes an efficient conversion algorithm for G.729 and G.723.1 speech codecs to reduce computational complexity of the communications between the G.729 and G.723.1 speech codecs. The proposed transcoding method incorporates four processes: line spectral pair (LSP) interpolation, pitch conversion, fast adaptive-codebook search, and fast fixed-codebook search. To reduce search comput...
متن کامل