Masanao Suzuki

Voice enhancement device by separate vocal tract emphasis and source emphasis

A voice intensifier capable of reducing abrupt changes in the amplification factor between frames and realizing excellent sound quality with less noise feeling by dividing input voices into the sound source characteristic and the vocal tract characteristic, so as to individually intensify the sound source characteristic and the vocal tract characteristic and then synthesize them before being output. The voice intensifier comprises a signal separation unit for separating the input sound signal into the sound source characteristic and the vocal tract characteristic, a characteristic extraction unit for extracting characteristic information from the vocal tract characteristic, a corrective vocal tract characteristic calculation unit for obtaining vocal tract characteristic correction information from the vocal tract characteristic and the characteristic information, a vocal tract characteristic correction unit for correcting the vocal tract characteristic by using the vocal tract characteristic correction information, and a signal synthesizing means for synthesizing the corrective vocal tract characteristic from the vocal tract characteristic correction unit and the sound source characteristic, so that the sound synthesized by the signal synthesizing means is output.

Speech coding translation for IP and 3G mobile integrated network

As a novel technique for IP and 3G mobile integrated networks, we propose a speech coding translation scheme that can improve the speech quality of intercommunication among different speech coding standards, such as G.729 of VoIP and AMR of IMT-2000. Through the direct conversion of transmission parameters, this scheme improves speech intelligibility with reduced processing delay compared with conventional schemes, which consist of auxiliary decoding and asynchronous re-encoding. A subjective test by 40 non-experts shows that the proposed scheme significantly improves the subjective speech quality by a probability of 99%. And processing delay in one-way transmission is reduced by 25 ms.

Efficient PSI-CELP DSP implementation

Pitch Synchronous Innovation-Code Excited Linear Prediction (PSI-CELP) algorithm has been adopted as a standard speech codec in the new Japanese Personal Digital Cellular (PDC) system. In this paper, we present an efficient implementation scheme for the PSI-CELP algorithm with a Digital Signal Processor (DSP). A PSI-CELP algorithm features the PSI technique, which can improve speech quality to repeat a part of a stochastic codeword with pitch period. We propose a specialized function that can support DSP instructions to allow efficient PSI implementation. According to our practical estimation with the PSI-CELP standard specification, our method can greatly reduce the costs for both parts complexity and required memory space.