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In this paper, cepstral smoothing is introduced to reduce musical noise in speech separated using computational auditory scene analysis (CASA). Our post-processing algorithm is composed of three steps. First, speech and interference are separated from noisy speech. Next, binary masks in the time-frequency domain are obtained using separated speech and noise signals. Finally, a mask smoothing algorithm which preserves speech onsets and quasi-stationary narrow-band structures is used to eliminate excessive peaks in time-frequency domain that lead to musical noise. Using an objective speech quality measure, we show that proposed method can improve speech quality.

A unit controlling the volume of audio system in an automobile is designed. Main component in the unit is TMS320C54. We realize an adaptive algorithm with the chip to distinguish noise from outside and signal from the audio system. And the volume is controlled by signal to noise ratio (SNR). One microphone should be located somewhere in the automobile. The unit picks up the electrical signal from the microphone and electrical signals at all speakers in the audio system. Multi-channel signal are considered in the adaptive algorithm. We should decrease the complexity of the algorithm as low as possible with regard to our specific problem so that a TMS320C54 chip can implement the algorithm.

Using SIP (session initiation protocol), a CTI system is proposed for providing real-time and all-round service in call center. Compared with the traditional call center, for our CTI system adopts SIP as its signaling protocol, which can integrate other successful Internet service in a simple measure, such as HTTP and SMTP. The customer can be served through main kinds of convenient ways. Besides, call center becomes more flexible.

This article presents an implementation of PSTN media gateway. The architecture of media gateway and the media processing scheme are described. It's designed from the idea of decomposed gateway. SIP, as a new VoIP technology, is adopted as the signaling protocol in the NGN. In the scheme we present the architecture of the hardware, the software control modules, and some other kernel technologies used in the media gateway.

This paper provides several research results about a technique of high quality telephony over LAN. Firstly, the value of the technique is discussed. Secondly, the delay performance of Ethernet LAN is tested by experiments both for the network layer and the application layer. Thirdly, a software architecture orientated to a real-time process like speech communication is proposed against the major reason of the overall transmission delay. Finally, as a conclusion, the possibility of realizing high quality telephony over LAN is demonstrated.