Packet-based embedded encoding for transmission of low-bit-rate-encoded speech in packet networks

摘要

Missing packets due to network congestion can seriously degrade the quality of received speech in packet voice communication. A variety of techniques exist that attempt to reconstruct the missing speech segments but these methods cannot be directly applied to low-bit-rate-encoded speech, such as ADPCM, where the decoded output depends not only on current transmitted codewords but also on the past history of the decoder. In this case, decoder parameters at the receiver lose track of encoder parameters after a missing packet leading to propagation of the decoding error into the following packet. A novel packet-based embedded encoding scheme is described that mitigates the effects of parameter mistracking caused by packet loss. The method may be applied to any low-bit-rate encoder that uses past information to decode current samples but specific results are given for its application to the CCITT 32 kbit/s ADPCM encoding standard. It is shown that packet-based embedded encoding of speech can double the maximum tolerable missing packet rate.