فهرست مطالب

Majlesi Journal of Multimedia Processing
Volume:1 Issue: 1, Mar 2012

  • تاریخ انتشار: 1391/02/23
  • تعداد عناوین: 6
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  • Mansour Sheikhan Page 1
    Low Delay-Code Excited Linear Prediction (LD-CELP) is considered as an attractive algorithm in speech coding domain. This algorithm was adopted by the International Telephone and Telegraph Consultative Committee (CCITT) for the coding of speech at 16 kbps with toll quality. However, operation at transmission rates lower than 16 kbps is desirable so that traffic can be accommodated during system overload conditions. In this paper, a 12.8 kbps LD-CELP algorithm is proposed in which codebook search module employs a network of Self-Organizing Maps (SOMs) to determine the optimum index value of shape codebook. Based on the occurrence frequency characteristics of codevectors, the 6 bits for shape codebook and 2 bits for gain codebook are used in this work to produce a 12.8 kbps coder. The performance comparison of the proposed neural codebook search module in the structure of 12.8 kbps LD-CELP with a conventional implementation of LD-CELP coder shows that execution time of the algorithm is reduced significantly. However, the degradation of voice quality in terms of Perceived Evaluation of Speech Quality (PESQ) and segmental Signal to Noise Ratio (SNRseg) is not significant.
  • Ehsan Lotfi Page 9
    In this paper, novel method based on tree representation of video shots is proposed to retrieve soccer events. The first phase consists of extracting suitable features and key frames from video shots. The features include: types of view, slow-motion and caption detection, whistle, player gathering and referee extraction. Then, using the tree representation of soccer shots the events are extracted. The proposed system is an extended method in multimodal representation and the retrieval process. Experimental results show high accuracy of proposed feature extraction algorithms and satisfying retrieval process.
  • Ahmad Hatam, Ali M. Doost Hoseini Page 15
    A rate allocation (RA) method in joint source/channel coding for transmission of images over varying channels with Markov model is proposed, in which the dependency of channel states during transmission of image packets is considered. For rate allocation (RA) problem, a weighted packet decoding error probability (PDEP) approximation method is presented. Our simulations show that the proposed RA method leads to a better performance over existing RA policies.
  • Omid Sharifi, Tehrani Page 23
    This paper aims to design and implement an algorithm for implementation of Simple Speech Recognition System based-on Digital Filters in AVR Microcontroller. This speech Distinction Algorithm presented here is used in automobiles, reinforcement Robots or in the disable person’s chair for controlling the chair through the user’s speaking. The necessary processes are done through using S-REC Algorithm and microcontroller MEGA-AVR instead of DSP processors. Microcontroller memory overflow problem is solved by using the proposed algorithm as results shows.
  • Mohsen Ashourian, Mohammad Reza Razmi Page 27
    In this paper we investigate several selective video encryption systems. Our proposed system works based on modification of the video content in discrete cosine transform domain. We select some of the main DCT coefficients and xor them with the key, also we do a permutation on the result.
  • Venu Madhav Kotturu Page 35
    Applications such as hands-free telephony, tele-classing and video-conferencing require the use of an acoustic echo canceller (AEC) to eliminate acoustic feedback from the loudspeaker to the microphone. Room acoustic echo cancellation typically requires adaptive filters with thousands of coefficients. Transform domain adaptive filter finds best solution for echo cancellation as it results in a significant reduction in the computational burden. Literature finds different orthogonal transform based adaptive filters for echo cancellation. In this paper, we present Hirschman Optimal Transform (HOT) based adaptive filter for elimination of echo from audio signals. Simulations and analysis show that HOT based LMS adaptive filter is computationally efficient and has fast convergence compared to LMS, NLMS and DFT-LMS. The computed Echo Return Loss Enhancement (ERLE), the general evaluation measure of echo cancellation, established the efficacy of proposed HOT based adaptive algorithm. In addition, the spectral flatness measure showed a significant improvement in canceling the acoustic echo. To achieve the goal of hardware implementation, which provides high performance, low power and reliability, we propose a VHDL model for a HOT Adaptive filter.