School of Information and Communications, Gwangju Institute of Science and Technology, Gwangju 500-712, Korea.
Sensors (Basel). 2011;11(9):8469-84. doi: 10.3390/s110908469. Epub 2011 Aug 31.
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.
本文提出了一种自适应冗余语音传输 (ARST) 方法，旨在提高无线多媒体传感器网络 (WMSN) 上语音流应用的感知语音质量 (PSQ)。该方法从接收的语音数据中估计 PSQ 和分组丢失率 (PLR)。随后，它决定是否需要传输冗余语音数据 (RSD)，以帮助语音解码器对高 PLR 下丢失的语音信号进行重建。根据决策，所提出的 ARST 方法控制 RSD 传输，然后优化语音编码的比特率，以对当前语音数据 (CSD) 和 RSD 比特流进行编码，以在分组丢失情况下保持语音质量。使用自适应多速率-窄带 (AMR-NB) 语音编解码器和 ITU-T 建议书 P.563 分别作为可扩展语音编解码器和 PSQ 估计，验证了所提出的 ARST 方法的有效性。实验表明，在 WMSN 中，采用所提出的 ARST 方法的语音流应用程序在分组丢失情况下显著提高了语音质量。
