Improving the transcoding capability of speech coders

Abstract

With the trend of merging various communication networks, a need arises to provide transcoding between different speech coding formats. Presently this means cross tandeming the two coders in each case. This results in both quality loss and extra delay. A possible alternative is using a bitstream mapping approach that directly converts parameter values. For several standard coders having a similar coding structure, it should be possible to generate comparable or better quality without adding much delay or complexity. This paper proposes a bitstream mapping method between ITU-T Recommendation G.729 and TIA IS-641. Informal listening tests and the perceptual subjective quality measure (PSQM) scores show that the proposed method has better quality than the cross tandeming method, while it has at least 5 ms less delay and six times less computation.

Frequency-Domain Adaptive Algorithm for Network Echo Cancellation in VoIP

“…As network systems migrate from traditional voice telephony over public switch telephone network (PSTN) to packetswitched networks for VoIP, improving the quality of services (QoS) for VoIP has been and will remain a challenge [1,2]. As described in [1], several factors that can affect the QoS for VoIP include the choice of speech coder-decoders (codecs) [3], algorithmic processing delay [4], and packet loss [5], where the algorithmic delay is one of the significant factors for determining the budget for delay introduced by network echo cancellers. The problem of network echo is introduced by the impedance mismatch between the 2-and 4-wire circuits of a network hybrid [6], which occurs in VoIP systems, where analog phones are involved in PCto-phone or phone-to-phone connections [7], where "PC" represents all-digital terminals.…”