Abstract: The present invention relates to the decoding/playback part of received sound data packets in systems for transmission of sound over packet switched networks. According to the invention, the lengths of received signal frames are manipulated by performing time expansion or time compression of one or more signal frames at time varying intervals and with time varying lengths of the expansion or the compression, said intervals and said lengths being determined so as to maintain a continuous flow of signal samples to be played back.

Abstract: The present invention relates to the decoding-/playback part of received sound data packets in systems for transmission of sound over packet switched networks. According to the invention, the lengths of received signal frames are manipulated by performing time expansion or time compression of one or more signal frames at time varying intervals and with time varying lengths of the expansion or the compression, said intervals and said lengths being determined so as to maintain a continuous flow of signal samples to be played back.

Abstract: The present invention relates to the decoding/playback part of received sound data packets in systems for transmission of sound over packet switched networks. According to the invention, the lengths of received signal frames are manipulated by performing time expansion or time compression of one or more signal frames at time varying intervals and with time varying lengths of the expansion or the compression, said intervals and said lengths being determined so as to maintain a continuous flow of signal samples to be played back.

Abstract: The present invention relates to echo suppression in a telephony system wherein a communication link is established over an analogue network and a digital network. The disclosed echo suppression invention provides cost-effective ways of minimizing an echo arising in the public switched telephone network (PSTN) by calculating a first value for the energy level of the signal from the analogue to the digital network, and likewise calculating a second value for the energy level of the signal from the digital to the analogue network. The sound from the analogue network to digital network is transmitted in dependence of the results of a comparison between the first and second values.

Abstract: Frequency domain nonlinear processing techniques are applied in the context of echo suppression. According to exemplary embodiments, the filtering characteristic of a nonlinear processor is dynamically adjusted by comparing the power spectrum of a residual echo component of a communication signal with that of the overall communication signal itself. For example, the filtering characteristic can be adjusted so that the nonlinear processor blocks only those signal frequencies in which the residual echo is, in some sense, dominant. The disclosed techniques make full duplex communication possible during the entirety of a conversation, even while residual echo is being actively suppressed.

Abstract: The present invention relates to the decoding-/playback part of received sound data packets in systems for transmission of sound over packet switched networks. According to the invention, the lengths of received signal frames are manipulated by performing time expansion or time compression of one or more signal frames at time varying intervals and with time varying lengths of the expansion or the compression, said intervals and said lengths being determined so as to maintain a continuous flow of signal samples to be played back.

Abstract: Adaptive gain control techniques provide correctly adjusted audio signal levels during the entirety of an Internet telephony conversation and are resilient to background noise and loudspeaker echo. Disclosed techniques can account for multiple near-end speakers, as well as changes in near-end environment. In an exemplary embodiment, an adaptive gain controller includes a gain control processor configured to adjust an analog gain for a microphone output signal based on measurements of the microphone output signal and on measurements of a loudspeaker input signal.

Abstract: Noisy speech parameters are enhanced by determining a background noise power spectral density (PSD) estimate, determining noisy speech parameters, determining a noisy speech PSD estimate from the speech parameters, subtracting a background noise PSD estimate from the noisy speech PSD estimate, and estimating enhanced speech parameters from the enhanced speech PSD estimate.