Abstract
Adaptive muting method using an optimized parametric shaping function as a part of the ITU-T G.722 Appendix IV packet loss concealment algorithm is proposed. The packet loss concealment algorithm incorporating an adaptive muting scheme is known to prevent the generation of unpleasant sounds during packet loss concealment. However, original muting uses a piece-wise linear muting curve according to packet errors so that our muting approach uses non-linear parametric shaping functions including sigmoid and raised-cosine. Training is substantially performed to determine the parameters of the two shaping functions in terms of objective speech quality measures, and optimal parameters are finally selected by subjective speech quality. Through extensive experiments, this proposed muting technique turns out to improve the performance of the reference muting mechanisms in the packet loss concealment algorithm of the G.722 Appendix IV under various experimental conditions.
Highlights
A variety of voice communication services through the internet have been building a growing interest in voice over internet protocol (VoIP) systems
VoIP applications are basically developed through a packet-based system over IP networks, which operate with the help of standard codecs such as ITU-T G.722, G.729, G.723.1, and adaptive multi-rate (AMR) [1]
VoIP applications have a critical problem of packet loss due to delay and jitter during the transmission of the speech data so that the quality of service (QoS) cannot be acceptable in poor network conditions [3]
Summary
A variety of voice communication services through the internet have been building a growing interest in voice over internet protocol (VoIP) systems. VoIP applications are basically developed through a packet-based system over IP networks, which operate with the help of standard codecs such as ITU-T G.722, G.729, G.723.1, and adaptive multi-rate (AMR) [1]. There are several methods for the sender-based reconstruction schemes including packet retransmission [6], interleaving [7], and sending error correction bits in voice packets using forward error correction (FEC) technique [8]. These methods require considerable increase in bandwidth, longer endto-end delay, or may require modifications on the sender side [9]. The PLC algorithm in enhanced voice services (EVS) codec [10], recently standardized by 3rd generation partnership project (3GPP), transmits side information such as pitch lag to the decoder side so it requires a special transmission format from the encoder at specific encoding modes [11]
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