Abstract: A system and method for processing of audio and speech signals is disclosed, which provide compatibility over a range of communication devices operating at different sampling frequencies and/or bit rates. The analyzer of the system divides the input signal in different portions, at least one of which carries information sufficient to provide intelligible reconstruction of the input signal. The analyzer also encodes separate information about other portions of the signal in an embedded manner, so that a smooth transition can be achieved from low bit-rate to high bit-rate applications. Accordingly, communication devices operating at different sampling rates and/or bit-rates can extract corresponding information from the output bit stream of the analyzer. In the present invention embedded information generally relates to separate parameters of the input signal, or to additional resolution in the transmission of original signal parameters.

Abstract: A modular system and method is for displaying and controlling a host computer application interface using a real-time video and control stream to a remote digital media receiver client is disclosed. The continuous graphical display pixels and audio of a host computer application interface window are divided into time segments of a predetermined length. These segments are then subsequently coded as video and audio streams. These video and audio streams are then sent to a remote digital media receiver client where they are decoded for display on a television device or played back on an audio system. the host computer application interface is controlled by user commands that originate at the remote digital media receiver client. The commands are sent over a control channel that is established between the host computer and the digital media receiver.

Abstract: A system and method are provided for processing audio and speech signals using a pitch and voicing dependent spectral estimation algorithm (voicing algorithm) to accurately represent voiced speech, unvoiced speech, and mixed speech in the presence of background noise, and background noise with a single model. The present invention also modifies the synthesis model based on an estimate of the current input signal to improve the perceptual quality of the speech and background noise under a variety of input conditions. The present invention also improves the voicing dependent spectral estimation algorithm robustness by introducing the use of a Multi-Layer Neural Network in the estimation process. The voicing dependent spectral estimation algorithm provides an accurate and robust estimate of the voicing probability under a variety of background noise conditions. This is essential to providing high quality intelligible speech in the presence of background noise.

Abstract: Systems and methods for creating a Universal Media Server (UMS) and associated Digital Media Renders (DMRs) are disclosed. This system provides functions such as in-home wired or wireless network for media distribution with PVR functions from a distributed archive while at the same time including functions such as an answering machine, voice recorder, voice over IP gateway, firewall, NAT, DHCP, and security monitoring. A variety of advanced features and functions are also disclosed.