An adaptive redundant speech transmission (ARST) approach to improve the perceived speech quality (PSQ) of speech streaming applications over wireless multimedia sensor networks (WMSNs) is proposed in this paper. The proposed approach estimates the PSQ as well as the packet loss rate (PLR) from the received speech data. Subsequently, it decides whether the transmission of redundant speech data (RSD) is required in order to assist a speech decoder to reconstruct lost speech signals for high PLRs. According to the decision, the proposed ARST approach controls the RSD transmission, then it optimizes the bitrate of speech coding to encode the current speech data (CSD) and RSD bitstream in order to maintain the speech quality under packet loss conditions. The effectiveness of the proposed ARST approach is then demonstrated using the adaptive multirate-narrowband (AMR-NB) speech codec and ITU-T Recommendation P.563 as a scalable speech codec and the PSQ estimation, respectively. It is shown from the experiments that a speech streaming application employing the proposed ARST approach significantly improves speech quality under packet loss conditions in WMSNs.
A sanitary survey was conducted to evaluate the water quality and mussel (Mytilus edulis) conditions of two administrative shellfish growing waters: those designated as shellfish growing water for export, and adjacent waters on the east coast of Changseon Island, Namhae, Korea. In all, 1,656 seawater and 166 mussel samples were collected at 46 stations for seawater and five stations for the shellfish from January 2007 to December 2009. Both seawater and mussels were examined for total coliforms and fecal coliforms. The standard plate count and most probable number of Escherichia coli were also determined for the shellfish samples. The range of the geometric means and the estimated 90th percentiles of fecal coliform for seawater samples at each station were <1.8 4
In this paper, an adaptive speech streaming method is proposed to improve the perceived speech quality (PSQ) of voice over wireless multimedia sensor network (WMSNs). First of all, the proposed method estimates the PSQ of the received speech data under different network conditions that are represented by the packet loss rates (PLRs). Simultaneously, the proposed method classifies the speech signal as either an onset or a nononset frame. Based on the estimated PSQ and the speech class, it determines an appropriate bit rate for the redundant speech data (RSD) that are transmitted with the primary speech data (PSD) to help reconstruct the speech signals of any lost frames. In particular, when the estimated PLR is high, the bit rate of the RSD should be increased by decreasing that of the PSD. Thus, the bandwidth of the PSD is changed from wideband to narrowband, and an artificial bandwidth extension technique is applied to the decoded narrowband speech. It is shown from the simulation that the proposed method significantly improves the decoded speech quality under packet loss conditions in a WMSN, compared to a decoder-based packet loss concealment method and a conventional redundant speech transmission method.
A packet loss concealment (PLC) algorithm is proposed to improve the quality of decoded speech when packet losses occur in a wireless sensor network. The proposed algorithm is mainly based on artificial bandwidth extension (ABE) from narrowband to wideband. It consists of three main functions: packet loss concealment in the narrowband, ABE in the modified discrete cosine transform (MDCT) domain, and smoothing of wideband MDCT coefficients with those of the last good frame. The performance of the proposed PLC algorithm is implemented by replacing the PLC algorithm employed in the ITU-T Recommendation G.729.1. The experimental results show that the proposed PLC algorithm provides significantly better speech quality than the PLC in the ITU-T G.729.1.
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