Rafi Rabipour

Speech bandwidth extension method and apparatus

A speech bandwidth extension method and apparatus analyzes narrowband speech sampled at 8 kHz using LPC analysis to determine its spectral shape and inverse filtering to extract its excitation signal. The excitation signal is interpolated to a sampling rate of 16 kHz and analyzed for pitch control and power level. A white noise generated wideband signal is then filtered to provide a synthesized wideband excitation signal. The narrowband shape is determined and compared to templates in respective vector quantizer codebooks, to select respective highband shape and gain. The synthesized wideband excitation signal is then filtered to provide a highband signal which is, in turn, added to the narrowband signal, interpolated to the 16 kHz sample rate, to produce an artificial wideband signal. The apparatus may be implemented on a digital signal processor chip.

Tandem-free VoIP conferencing: a bridge to next-generation networks

This article surveys approaches to teleconferencing in voice over IP networks. The considerations for conferencing include perceived quality, scalability, control, and compatibility. Architectures used for conferencing range from centralized bridges to full mesh. Centralized conference bridges used with compressed speech degrade speech quality when multiple talkers are mixed and subjected to tandem coding operations. Full mesh and multicast solutions (mixing at the end-points) are inappropriate when the number of conferees is large. This article discusses a hybrid solution that incorporates tandem-free bridging (the bridge selects and forwards packets) and endpoint mixing.

Speaker selection for tandem-free operation VoIP conference bridges

A conventional voice-over-internet protocol (VoIP) conference bridge reduces the speech quality due to tandeming the mixed multi-speaker signal with high compression speech codecs. One solution is to select and forward the compressed signal(s) to the endpoints, where they are decoded and mixed. In such arrangements, speaker selection is usually accomplished with an order-based approach which prevents listeners from interrupting the current speaker(s). This paper presents an alternative in which talking privileges are assigned based on order of activity and signal power. Subjective evaluations indicate that speaker switching is smooth, nearly transparent, and unanimously preferred over a VoIP conference with tandemed connections.

Tandem-free operation for VoIP conference bridges

Traditional telephone conferencing has been accomplished by way of a centralized conference bridge. The tandem arrangement of high compression speech codecs in conventional VoIP conference bridges lead to speech distortions and require a substantial number of computations. Decentralized architectures avoid the speech degradations and delay, but lack strong control and depend on silence suppression to make the endpoint bandwidth and processing requirements scalable. One solution is to use centralized speaker selection and forwarding, and decentralized decoding and mixing. This approach eliminates the problem of tandem encodings but maintains centralized control, thereby improving the speech quality and scalability of the conference. This paper considers design options and solutions for this model in the context of modern IP telephony networks. Performance was evaluated with real conferees over live conferences using a PC-based bridge and a third-party endpoint. Conferees strongly preferred the speech quality of the new arrangement to that of a conventional VoIP conference bridge.