Nortel Networks Corporation

Mitel Networks Corporation

Zarlink Semiconductor Limited

Press-ir Inc.

Botrus Teleconferencing & Acoustics Consulting, Ltd.

Rafik Alphonse Goubran

Dieter Schulz

Mohamed Samy El-Hennawey

Graham Thompson

Zhihong Qian

Charn Leung (David) Lo

Ayman M Radwan

Lijing Ding

Andre John Van Schyndel

Radamis Botros

Frank-Dietrich Knoefel

Marco Nasr

Ying Huang

Jeff Lariviere

Raymond Bruce Wallace

Mohamed Samy El-Hennaway

Charn Leung Lo

System and method that use a pressure sensitive mat to monitor vitals signs of an individual positioned on the pressure sensitive mat. The pressure sensor mat is coupled to pressure sensors that each have an adjustable pressure responsivity. In order to be able to monitor the vital signs without interruption due to the individual repositioning themself on the pressure sensitive mat and thereby running the risk of increasing the pressure on some pressure sensors to the point of saturation, or running the risk of decreasing the pressure on other pressure sensors to the point losing any usable signal, the system comprises circuitry that monitors a baseline pressure sensor at one pressure sensor and controls, as a function of the monitored signal, the pressure responsivity at a neighbor pressure sensor. This allows the neighbor sensor to generate a monitored signal with discernable time dependent features associated with the individual's vital signs.

Pressure sensitive mat system with dynamically calibratable pressure sensors and method for non-obtrusive monitoring of vital signs and other health metrics

An apparatus (), method, and computer program to assess VoIP speech quality () using access to degraded signals is provided. Different types of impairment () have different effect, on speech quality. Preferred embodiments address up to four different types of impairment that affect VoIP signal quality: packet loss (), speech clipping in time (), noise () and echo. An overall assessment algorithm factors in degradation due to various impairment factors to generate an overall speech quality assessment score or value.

Method and apparatus for non-intrusive single-ended voice quality assessment in VoIP

Transcoders and mixers having reduced algorithmic delay and processing complexity. An improved mixer for signals having encoded speech parameters wherein the parameters obtained through decoding are used by a parameter estimator to improve the encoding by providing a parameter estimate for the mixed signal. In the case of pitch parameters, the mixer uses the principle of strong-pitch-domination. The mixing of wideband signals is simplified by performing mixing of individual lower and upper sub-bands. A transcoder and a mixer that converts a wideband signal into a narrowband signal relies upon high frequency suppression. A transcoder and a mixer that converts a narrowband signal into a wideband signal relies upon filter combination.

Transcoders and mixers for voice-over-IP conferencing

An apparatus (), method, and computer program to assess VoIP speech quality () using access to degraded signals. Different types of impairment () have different effect on speech quality. Preferred embodiments address two distinct types of impairment that are unique to VoIP: packet loss () and speech clipping in time ().

A method is provided for discriminating between the case of a single talker with an acoustic reflection and the case of two talkers, regardless of their power levels. The method is implemented in real time by performing a cross-correlation between pairs of average power signals originating from pairs of beamformers. A detection decision is then made based on the value of the cross correlation and its lag.

Detecting acoustic echoes using microphone arrays

A method of self-discovery and self-calibration is provided for allowing arbitrary placement of audio and video components in a multimedia conferencing system. In particular, one or more markers are provided on the audio components (e.g. microphone arrays, etc.) that are detectable by the video components (e.g. cameras). A unique signature (e.g. flashing sequence, color, etc.) characterizes each marker so that its exact location relative to the camera may be calculated. A self-calibration operation is then performed to relate, regulate and standardize dimensions and locations in the conferencing environment to the video system.

System and method of self-discovery and self-calibration in a video conferencing system

An insertion-based error concealment method and apparatus are provided whereby, instead of directly inserting white noise, a filter is created to shape the white noise. The filtered white noise is then used to replace lost data. The method of the present invention is implemented by first estimating the power spectrum of the previous frame; then designing a filter with transfer function H(f), where |H(f)|=the estimated power spectrum; and finally generating the replacement packet using noise which has been spectrally modified by the filter. The resulting filtered noise has the same power spectrum as the previous packet but is not highly correlated with it.

Packet loss compensation method using injection of spectrally shaped noise

Handsfree telephony continues to be an increasingly desirable feature of modern telecommunications, whether in a conference room or mobile setting. Fundamental to the user acceptability of these systems is the performance of algorithms for acoustic echo cancellation, the purpose of which are to prevent the far-end signal from being transmitted back to the far-end talker. Most speech coding algorithms are based on some variant of Linear Predictive Coding (LPC), which reduces the amount of bits sent across a channel. Instead of doing echo cancellation in the time domain, the invention involves operating an acoustic echo canceller on the LPC parameters at the receiver, before the decoding stage.

Linear predictive coding based acoustic echo cancellation

The invisible acoustic screen for reducing sound leakage from handsfree telephones, loudspeaking computer terminals, and the like in a workspace includes a sound reducing apparatus comprising: a sound producing loudspeaker; a cancelling loudspeaker; a sensing microphone having an output; a delay circuit; and, an adaptive filter circuit having a value determined by a transfer function between the sound producing loudspeaker and the sensing microphone and by a transfer function between the cancelling loudspeaker and the sensing microphone; and the output of the sensing microphone being fed back to the adaptive filter circuit; and the received signal coupled through a delay circuit to the sound producing loudspeaker.

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Ottawa, Canada

Rafik Alphonse Goubran