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Dual-rate G.723.1 speech coder has been widely applied to real-time video and teleconferencing applications where reduced bandwidth and good voice quality is required. This paper presents an efficient implementation of G.723.1 speech coder. To simplify the excitation quantization procedure which is the most computationally demanding, we propose fast(More)
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(More)
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 adaptive-codebook search. For maintaining minimum distortion,(More)
This paper proposes a novel speaker identification system based on score fusion of various resolution filterbanks. The proposed system uses multiple features which are extracted from filterbanks having various spectral resolutions. Each speaker model is constructed by independent feature set, but the system makes final decision by combining the outcome of(More)
The paper describes a preliminary experiment to find a supplementary feature for speaker verification. In conventional speaker verification systems, spectral features, such as the Mel frequency cepstral coefficient (MFCC), are used universally for all speakers. For some specific applications demanding high security, however, the system also needs to adopt(More)
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