Abstract
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.
Highlights
wireless sensor networks (WSNs) have led to another innovation, wireless multimedia sensor networks (WMSNs), which interconnect sensor nodes equipped with multimedia devices such as cameras and microphones [2]
In order to demonstrate the effectiveness of the proposed adaptive redundant speech transmission (ARST) approach, a speech streaming application was first implemented by using the adaptive multirate-narrowband (AMR-NB) speech codec and ITU-T Recommendation
It was shown from the figure that the packet loss concealment (PLC) approach improved speech quality more than the fixed redundant speech transmission (RST) approach did under the packet loss rate (PLR) below 5%
Summary
Based on advanced technologies for low power and highly integrated digital electronics, wireless sensor networks (WSNs) have emerged and received significant attention as they provide numerous functional applications, e.g., environmental monitoring, human tracking, and military surveillance [1]. In order to improve the speech quality in speech streaming applications against packet losses, a number of error protection methods were proposed for IP networks. These methods are typically classified into receiver-based schemes and sender-based schemes, as shown in Figures 1 and 2, respectively [9]. The speech streaming capability over sensor nodes in an operational coal mine [3] was investigated by comparing two waveforms recovered by the receiver-based scheme and the sender-based scheme, respectively It was revealed in [3] that a speech streaming application employing the receiver-based scheme could accommodate a higher speech coding bitrate under a low packet loss rate (PLR) condition. A Speech Streaming Application Using the Proposed Adaptive Redundant Speech Transmission
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