This paper describes our work on real-time twoparty and multi-party VoIP (voice-over-IP) systems that can achieve high perceptual conversational quality. It focuses on the fundamental understanding of conversational quality and its trade-offs among the design of speech codecs and strategies for network control, playout scheduling, and loss concealments. We have studied three key aspects that ad...

Channel estimation in communication systems is one of the most important issues that can reduce the error rate of sending and receiving information as much as possible. In this regard, estimation of OFDM-based wireless channels using known sub-carriers as pilot is of particular importance in frequency domain. In this paper, channel estimation under short cyclic prefix (CP) in OFDM system is con...

this study proposes a modified version of cultural algorithms (cas) which benefits from rule-based system for influence function. this rule-based system selects and applies the suitable knowledge source according to the distribution of the solutions. this is important to use appropriate influence function to apply to a specific individual, regarding to its role in the search process. this rule ...

A parallel algorithm for generating t-ary tree sequences in reverse B-order is presented. The algorithm generates t-ary trees by 0-1 sequences, and each 0-1 sequences is generated in constant average time O(1). The algorithm is executed on a CREW SM SIMD model, and is adaptive and cost-optimal. Prior to the discussion of the parallel algorithm a new sequential generation with O(1) average time ...

We present an integrated approach of full-band audio time scale modification for Voice over IP communication. The concept is based on a low complexity adaptive playout method that uses frame dropping and audio concealment for time shrinking and stretching, respectively. The existing version of this method is improved using a classifier that assists in choosing which audio frames can be dropped ...

In this paper we describe a novel adaptive method of speech quality control which may be used to adjust three call parameters: speech codec configuration, playout buffer size, and amount of FEC (Forward Error Correction) mechanism information during VoIP (Voice over Internet Protocol) call under changing network conditions. This solution differs from previously proposed because it utilizes audi...