PROBLEM TO BE SOLVED: To recognize the information of nurses to be changed by work shifts and the information of a portable terminal to which a calling operation is being performed for a call from a patient in a nurse call master unit. SOLUTION: In a memory provided in a controller, team and person-in-charge number information for which a nurse team and person-in-charge numbers attached to respective nurses are related, nurse and portable terminal information for which the names of the nurses and the calling numbers of the portable terminals are related, person-in-charge number and portable terminal number information for which the person-in-charge numbers and the calling numbers of the portable terminals are related, and slide calling information for sliding and calling the nurses further are preserved. In each patient information of an escutcheon displayed on the LCD 13 of a PC master unit, the name of the nurse in charge corresponding to the work shift of the nurse is displayed. Then, when a call is made from a bed slave unit, the information T1 of the part of patient's name "patient (a)" and the information T2 of the portable terminal 2 being called at present on the basis of the information stored in the memory and the nurse are popped up and displayed from the escutcheon. COPYRIGHT: (C)2006,JPO&NCIPI

Nurse call system

At least one exemplary embodiment is directed to a method and/or a device for voice operated control. The method can include method measuring an ambient sound received from at least one Ambient Sound Microphone, measuring an internal sound received from at least one Ear Canal Microphone, detecting a spoken voice from a wearer of the earpiece based on an analysis of the ambient sound and the internal sound, and controlling at least one voice operation of the earpiece if the presence of spoken voice is detected. The analysis can be a non-difference comparison such as a correlation analysis, a cross-correlation analysis, and a coherence analysis.

Method and Device for Voice Operated Control

Controlling the color temperature of a composite light source including at least one discrete-spectrum light source is disclosed. For example, the color temperature of a composite light source including at least one discrete-spectrum light source may be determined and/or adjusted based on one or more of the ambient color temperature of a space, the actual temperature of the space, the relative brightness of the space, the occupancy of the space, a time clock, a demand response command (e.g., from an electrical utility), the absolute location of the composite light source, the location of the composite light source relative to other light sources, inputs from a camera or other external devices, the operation of appliances or other machines in the vicinity of the composite light source, media content being utilized in the vicinity of the composite light source, and/or other sensor inputs.

Systems and methods for controlling color temperature

A novel adaptive beamforming technique with enhanced noise suppression capability. The technique incorporates the sound-source presence probability into an adaptive blocking matrix. In one embodiment the sound-source presence probability is estimated based on the instantaneous direction of arrival of the input signals and voice activity detection. The technique guarantees robustness to steering vector errors without imposing ad hoc constraints on the adaptive filter coefficients. It can provide good suppression performance for both directional interference signals as well as isotropic ambient noise.

Robust adaptive beamforming with enhanced noise suppression

A sound zoom method, medium, and apparatus generating a signal in which a target sound is removed from sound signals input to a microphone-array by adjusting a null width that restricts a directivity sensitivity of the microphone array, and extracting a signal corresponding to the target sound from the sound signals by using the generated signal. Thus, a sound located at a predetermined position away from the microphone array can be selectively obtained so that a target sound is efficiently obtained.

Sound zoom method, medium, and apparatus

The adaptive beamformer unit (191) comprises: a filtered sum beamformer (107) arranged to process input audio signals (u l, u2) from an array of respective microphones (101, 103), and arranged to yield as an output a first audio signal (z) predominantly corresponding to sound from a desired audio source (160) by filtering with a first adaptive filter (fl (-t)) a first one of the input audio signals (u l) and with a second adaptive filter (f2(-t)) a second one of the input audio signals (u2), the coefficients of the first filter (fl(-t)) and the second filter (f2(-t)) being adaptable with a first step size (al) and a second step size ((x2) respectively; noise measure derivation means (111) arranged to derive from the input audio signals (ul, u2) a first noise measure (xl) and a second noise measure (x2); and an updating unit (192) arranged to determine the first and second step size (al, (x2) with an equation comprising in a denominator the first noise measure (xl) for the first step size (al), respectively the second noise measure (x2) for the second step size (a2). This makes the beamformer relatively robust against the influence of correlated audio interference. The beamformer may also be incorporated in a sidelobe canceller topology yielding a more noise cleaned desired sound estimate, which can be used in a related, more advanced adaptive filter (fl (-t), f2(-t)) updating. Such a beamformer is typically useful for application in handsfree speech communication systems.

Adaptive beamformer, sidelobe canceller, handsfree speech communication device

The relatively robust adaptive beamformer, comprises: a filtered sum beamformer (107) to process input audio signals (ul, u2, u3) from an array of respective microphones (101, 103, 105), and arranged to yield as an output a first audio signal (z) predominantly corresponding to sound from a desired audio source (160); and a noise estimation e.g. when incorporated in a sidelobe canceller topology an adaptive noise estimator (150), arranged to derive a noise signal (y) which is subtracted from the first audio signal (z) to obtain a noise cleaned second audio signal (r), and further comprises a scaling factor determining unit (170) arranged to provide a scale factor (S) as a function of a ratio (Q) of the sidelobe canceling, and being arranged to scale the adaptation step size with the scale factor (S), so that the sidelobe canceller only adapts quickly if it is relatively well locked on the desired audio source, but is rather insensitive to interference from noise sources.

Adaptive beamformer with robustness against uncorrelated noise

A sound source location is estimated by particle filtering where a set of particles represents a probability density function for a state variable comprising the sound source location. The method includes determining the weight for a particle in response to a correlation between estimated acoustic transfer functions from the sound source to at least two sound recording positions. A weight update function may specifically be determined deterministically from the correlation and thus the correlation may be used as a pseudo-likelihood function for the measurement function of the particle filtering. The acoustic transfer functions may be determined from an audio beamforming towards the sound source. The audio weight may be combined with a video weight to generate a multi-modal particle filtering approach.

Estimating a sound source location using particle filtering

A system (1) is described which is suited for suppressing audio distortion. The system comprises echo cancelling means g1, g2 coupled between an audio output (4) and a distorted desired audio sensing microphone array (3), and a filter arrangement (7) coupled to the echo cancelling means g1, g2 and/or the microphone array (3). The filter arrangements' coefficients represent reverberation distortion in the desired audio sensed by the microphone array (3).

Audio system having reverberation reducing filter

The present invention relates generally to the field of occupancy detection, and more particularly to an occupancy detection system and a corresponding method suitable for a presence controlled system. A method and system for determining occupancy of a room is provided, in which a signal from a movement sensor positioned in the room is analyzed to detect variations in the signal which correspond to movements of an occupant of the room. The room is determined to be empty if in the received signal a leave pattern is identified. The leave pattern is a predetermined pattern which has been associated with an occupant leaving the room. The method and system provide reliable occupancy detection in more or less static environments like for instance offices, and when utilized in for instance a presence controlled lighting system, the number of frustrating erroneous "switch-off" events of the light in a situation where a room is still occupied by occupants involved in relative still/static activities is then decreased.

Bahaa Eddine Sarroukh

Papers

Adaptive beamformer, sidelobe canceller, handsfree speech communication device

Adaptive beamformer with robustness against uncorrelated noise

Estimating a sound source location using particle filtering

Audio system having reverberation reducing filter

Occupancy detection method and system