Abstract: A turnstile comprising means for dictating a person entering or exiting the lane, a barrier positioned to move between a position blocking a lane and a position where it allows passage through the lane, under control of a motor; means for checking the credentials of a person attempting to pass through the lane and a brake, the turnstile having a mode of operation in which in the event that a user enters an exit side of the lane without presenting a credential, and an authorized person is not detected on the entry side of the lane, the brake is unlocked and the barrier is held in position in a closed mode but is able to be pushed or pulled by the person who has entered from the exit side in order for that person to exit; and the motor assists movement in the pushing or pulling direction by moving the barrier at a controlled speed, less than a speed at which it operates the barrier if the exiting user presents a credential.

Abstract: The present disclosure provides a method for adding realism to synthetic speech. The method includes receiving text (218) that is to be converted into synthetic speech from a mobile device (108). The text (218) may include embedded emoticons indicating a first prosody information and a predefined sound stored in a stored data repository (208). The method also includes identifying a user associated with the text (218) based on a comparison between metadata associated with the text (218) and user profiles stored in the stored data repository (208); retrieving a speech font from a speech data corpus associated with the user stored in the stored data repository (208). The speech font includes a second prosody information and a predefined accent of the user. The method further includes converting the text (218) into synthetic speech based on the retrieved speech font, which is being modulated based on the emoticon.

Abstract: A turnstile comprising means for dictating a person entering or exiting the lane, a barrier positioned to move between a position blocking a lane and a position where it allows passage through the lane, under control of a motor; means for checking the credentials of a person attempting to pass through the lane and a brake, the turnstile having a mode of operation in which in the event that a user enters an exit side of the lane without presenting a credential, and an authorised person is not detected on the entry side of the lane, the brake is unlocked and the barrier is held in position in a closed mode but is able to be pushed or pulled by the person who has entered from the exit side in order for that person to exit; and the motor assists movement in the pushing or pulling direction by moving the barrier at a controlled speed, less than a speed at which it operates the barrier if the exiting user presents a credential.

Abstract: The present disclosure relates to methods of systems for radiotherapy. Embodiments of the present disclosure may provide a carousel to display the plurality of images to a user and receive an input from the user for selecting one or more images displayed in the carousel. A graphical user interface may be provided for displaying to the user the one or more images selected from the carousel.

Abstract: Voice over internet protocol (VoIP) devices and conferencing systems may include a spatial encoder associated with a first endpoint and a spatial renderer associated with a second endpoint. The spatial renderer may configured to receive audio data. The audio data may be rendered among a plurality of speakers based on a first set of spatial information for a plurality of microphones associated with the first endpoint, and a second set of spatial information for the plurality of speakers associated with the second endpoint. A method for generating a sound field may include determining spatial information for a plurality of microphones in a local room, determining spatial information for a plurality of speakers in a remote room, mapping the spatial information for the plurality of microphones and the spatial information for the plurality of speakers, and generating a sound field in the remote room based on the mapping.

Abstract: A mass digitization system may include a work surface rotatably coupled to a support structure, and a motor coupled to the work surface to selectively rotate the work surface. An imaging station may be positioned proximate the work surface to capture digital images of items on a receiving surface of the work surface. The motor may rotate the work surface and the imaging station may include an imaging device to capture images of items on the receiving surface as the items are positioned in the image capture area of the imaging device.

Abstract: The present disclosure describes a system (100) for reducing background noise from a speech audio signal generated by a user. The system (100) includes a user device (102) receiving the speech audio signal, a noise reduction device (118) in communication with a stored data repository (208), where the noise reduction device is configured to convert the speech audio signal to text; generate synthetic speech based on the converted text; optionally determine the user as an actual subscriber based on a comparison between the speech audio signal with the synthetic speech; and selectively transmit the speech audio signal or the synthetic speech based on comparison between the predicted subjective quality of the recorded speech and the synthetic speech.

Abstract: The present disclosure provides a method for adding realism to synthetic speech. The method includes receiving text (218) that is to be converted into synthetic speech from a mobile device (108). The text (218) may include embedded emoticons indicating a first prosody information and a predefined sound stored in a stored data repository (208). The method also includes identifying a user associated with the text (218) based on a comparison between metadata associated with the text (218) and user profiles stored in the stored data repository (208); retrieving a speech font from a speech data corpus associated with the user stored in the stored data repository (208). The speech font includes a second prosody information and a predefined accent of the user. The method further includes converting the text (218) into synthetic speech based on the retrieved speech font, which is being modulated based on the emoticon.

Abstract: Embodiments of the present disclosure include an apparatus (116) configured to perform beamforming on multiple audio input signals. The apparatus (116) includes a first plurality of microphones (302, 502) configured to resolve first audio input signals within a first frequency range. The apparatus (116) also includes at least one microphone (504) configured to resolve second audio input signals within a second frequency range. A lowest frequency in the first frequency range is greater than a lowest frequency in the second frequency range.

Abstract: The present disclosure describes a system (100) for reducing background noise from a speech audio signal generated by a user. The system (100) includes a user device (102) receiving the speech audio signal, a noise reduction device (118) in communication with a stored data repository (208), where the noise reduction device is configured to convert the speech audio signal to text; generate synthetic speech based on the converted text; optionally determine the user as an actual subscriber based on a comparison between the speech audio signal with the synthetic speech; and selectively transmit the speech audio signal or the synthetic speech based on comparison between the predicted subjective quality of the recorded speech and the synthetic speech.

Abstract: This disclosure describes a system and method for negotiating video camera and display capabilities in a multi-camera/multi-display video conferencing environment. The method includes the at least one processor (120, 210, 218) determining negotiation information associated with a group of local devices (112). The negotiation information includes relative spatial information and data stream characteristics. The method also includes the at least one processor (120, 210, 218) defining the determined negotiation information in a predetermined format compatible with at least one remote device (114). The method further includes the at least one processor (120, 210, 218) applying the determined negotiation information for negotiating capability of at least one device (112-1) among the group of local devices (112) and the at last one remote device (114) for data communication.

Abstract: A conferencing system may comprise an electronic display configured to display remote video generated within a remote conference room, a speaker configured to reproduce remote audio generated within the remote conference room, and a processor configured to receive local audio generated within a local conference room and picked up by a microphone assembly that is part of a separate device from the electronic display. A related method may include displaying remote video on an electronic display of an all-in-one display, reproducing remote audio through at least one speaker of the all-in-one display; and performing echo cancellation of local audio using the remote audio as an echo cancellation reference. Another method may include receiving, at an all-in-one display, a plurality of local audio signals from a plurality of microphone assemblies that are separate from the all-in-one display, and controlling, within the all-in-one display, gating of the plurality of microphone assemblies.

Abstract: Embodiments include methods and apparatuses for echo cancelation involving multiple audio channels and the production, sensing, or a combination thereof of multiple audio channels in conferencing systems. A conferencing apparatus with a plurality of speakers configured to generate outgoing acoustic waves responsive to a multi-channel audio signal. One or more microphones are configured to sense incoming acoustic waves from the plurality of speakers and from locally produced acoustic waves from a participant of a conference to generate one or more incoming audio signals. A processor is operably coupled to the plurality of speakers and the one or more microphones. The processor is configured to perform acoustic echo cancelation on the one or more incoming audio signals relative to at least two different channels of the multi-channel audio signal.

Abstract: This disclosure describes a system that uses a virtual audio device system for unified communications (UC) applications between a host device and a plurality of audio devices (100, 130), which includes a host device (108), a plurality of audio devices (104, 106), and a virtual audio device system (124). The host device (108) executes a Unified Communications (UC) application (114) to generate a first audio data stream. The host device (108) is in communication with a microphone (110) and a speaker (112). The plurality of audio devices (104, 106) receive the first audio data stream generated by the host device, where each of the plurality of audio devices (104, 106) generates its own audio data stream. Further, the virtual audio device system (124) establishes a communication channel between the UC application (114) and the plurality of audio devices (104, 106) based on the plurality of audio devices (104, 106) connected to the host device (108).

Abstract: This disclosure describes a system (100, 130) for a USB to Bluetooth audio bridging method and apparatus. The system includes a host device (102) having a first universal serial bus (USB) interface (107) and a second USB interface (110); an adaptor (108) in connection with the first USB interface (107) which is compatible to operate based on a Bluetooth-type communication protocol; a peripheral device (106) coupled to the second USB interface (110); and a mobile communication device (104) linked to the host device (102) via the adaptor (108). The mobile communication device (104) communicates an audio data signal to the adaptor (108) and an audio bridging device (116). The audio bridging device (116) maps the second USB interface (110) to the adaptor (108) and routes audio data signals alone or in combination with another audio data signals to the peripheral device (106) via the second USB interface (110).

Abstract: This disclosure describes a system and method for negotiating video camera and display capabilities in a multi-camera/multi-display video conferencing environment. The method includes the at least one processor (120, 210, 218) determining negotiation information associated with a group of local devices (112). The negotiation information includes relative spatial information and data stream characteristics. The method also includes the at least one processor (120, 210, 218) defining the determined negotiation information in a predetermined format compatible with at least one remote device (114). The method further includes the at least one processor (120, 210, 218) applying the determined negotiation information for negotiating capability of at least one device (112-1) among the group of local devices (112) and the at last one remote device (114) for data communication.

Abstract: This disclosure describes a virtual multipoint control unit (112, 300) for unified communications. The virtual multipoint control unit (112, 300) communicates with multiple devices (102, 104) over a network (106). The control unit (112, 300) includes multiple unified communication (UC) applications (208, 314) being executed on the devices (102, 104). The UC applications (208, 314) decode received audio data stream and video data stream. A virtual imaging device (214) and a virtual audio device (216) are mapped to the UC applications (208, 314). A virtual video mixer (210) receives the decoded video data stream from the UC applications (208, 314) via the virtual imaging device (214). A virtual audio mixer (212) receives the decoded audio data stream from the UC applications (208, 314) via the virtual audio device (216).

Abstract: Embodiments of the present disclosure include an apparatus (210, 230, 240, 250) configured to perform beamforming on multiple audio input signals. The apparatus (210, 230, 240, 250) includes one or more illumination devices (222, 232, 242, 252) and a beamforming microphone system (116) integrated with the one or more illumination devices. The beamforming microphone system (116) includes a first plurality of microphones (302, 502) configured to resolve first audio input signals within a first frequency range. The beamforming microphone system (116) also includes at least one microphone (504) configured to resolve second audio input signals within a second frequency range. A lowest frequency in the first frequency range is greater than a lowest frequency in the second frequency range.

Abstract: The present disclosure relates to methods of systems for radiotherapy. Embodiments of the present disclosure may provide a carousel to display the plurality of images to a user and receive an input from the user for selecting one or more images displayed in the carousel. A graphical user interface may be provided for displaying to the user the one or more images selected from the carousel.

Abstract: A method of preparing an aqueous dispersion comprising silver particles of mean diameter from 0.5 to 25 nm by weight and an aqueous carrier liquid, including the steps of i) providing a mixture comprising at least one silver salt, aqueous carrier liquid and a stabiliser for the particles ii) contacting the mixture with a non-ionic or covalent reducing agent to form a reaction mixture iii) causing the at least one silver salt to react with the reducing agent to form a dispersion comprising silver particles and acid wherein step iii) is partly or completely performed in the presence of anion exchange resin whereby the acid is exchanged for a hydroxide ion from the resin and/or is sorbed by the resin.