Abstract: Methods, devices, and systems for coding and decoding audio are disclosed. Digital samples of an audio signal are transformed from the time domain to the frequency domain. The resulting transform coefficients are coded with a fast lattice vector quantizer. The quantizer has a high rate quantizer and a low rate quantizer. The high rate quantizer includes a scheme to truncate the lattice. The low rate quantizer includes a table based searching method. The low rate quantizer may also include a table based indexing scheme. The high rate quantizer may further include Huffman coding for the quantization indices of transform coefficients to improve the quantizing/coding efficiency.

Abstract: A camera tracking system includes a controllable camera, an array of microphones, and a controller. The microphones are positioned adjacent the controllable camera and are at least responsive to ultrasound emitted from a source. The microphones may additionally be capable of responding to sound in the audible spectrum. The controller receives ultrasound signals communicated from the microphones in response to ultrasound emitted from the source and processes the ultrasound signals to determine an at least approximate location of the source. Then, the controller sends one or more command signals to the controllable camera to direct at least approximately at the determined location of the source. The camera tracking system tracks the source as it moves and continues to emit ultrasound. The source can be an emitter pack having one or more ultrasonic transducers that produce tones that sweep form about 24-kHz to about 40-kHz.

Abstract: Methods, devices, and systems for coding and decoding audio are disclosed. At least two transforms are applied on an audio signal, each with different transform periods for better resolutions at both low and high frequencies. The transform coefficients are selected and combined such that the data rate remains similar as a single transform. The transform coefficients may be coded with a fast lattice vector quantizer. The quantizer has a high rate quantizer and a low rate quantizer. The high rate quantizer includes a scheme to truncate the lattice. The low rate quantizer includes a table based searching method. The low rate quantizer may also include a table based indexing scheme. The high rate quantizer may further include Huffman coding for the quantization indices of transform coefficients to improve the quantizing/coding efficiency.

Abstract: Systems, methods, and devices for providing an external wireless interface for use with a desktop telephone are disclosed. In accordance with various embodiments of the present disclosure, a telephone is provided which may include a universal connector operable to removably connect to any of a plurality of external radio transceivers. The external radio transceivers operate according a plurality of wireless communication protocols. In one embodiment, the universal connector may comprise a USB connector, and the external radio transceiver may be embodied as a USB dongle.

Abstract: According to the principles of the invention, there is provided a system, apparatus, and method for automated call launching in a call conferencing network. The system includes computer executable code for sharing information among a plurality of multipoint conference units, so that a remote user may, from a web site for example, initiate a conference call from one of the multipoint conference units. Using distributed code, a user's call request initiates a search of the network for a multipoint conference unit with adequate capacity and suitable capabilities for a requested call. Once a multipoint conference unit is selected, the multipoint conference unit calls out to selected participants who may accept or decline the call. This call setup is transparent to the user, who receives an acknowledgment once the conference call has been initiated.

Abstract: A system and method for optimal transmission of a multitude of video pictures consisting of end video encoders wherein more than one encoder is utilized. The system and method improve the image quality of a broadband image comprised of a plurality of images by generating a new bit rate encoder based upon a prior bit rate as determined by a bit rate determiner, thereby managing the distribution of available bandwidth.

Abstract: A conference link between devices in teleconference system at one site is disclosed. The linked devices include video conference units, speakerphones or stand-alone loudspeakers. Audio data can be transmitted through the conference link between devices. Audio data processing can be performed in either a video conference unit or a speakerphone. The microphones and loudspeakers in the video conference unit may be eliminated. The microphones and loudspeakers in the speakerphone are used. Other data, for example directories of dialing information, may also be transmitted through the conference link and the data processing may be allocated among processors in devices connected by conference links. The conference link may be wired or wireless, analog or digital. The devices may be linked by conference link in parallel or series. A dialing program can adapt the dialing stream with the locations of the near end site and the dialed far end site.

Abstract: A system and method for adjusting a video bit rate (VBR) over a network includes reducing the VBR if the network incurs a packet loss (PL) that is greater than a PL threshold; increasing the VBR if the PL is less than or equal to the PL threshold over a maximum integer number of time intervals; and increasing the maximum integer number of time intervals if the PL is greater than the packet loss threshold at the increased VBR. In addition, the VBR is increased over consecutive time intervals until a maximum video bit rate is restored, if the PL over each consecutive time interval is less than or equal to the packet loss threshold.

Abstract: A video teleconferencing method for providing reservationless conferencing. The method involves creating a conference profile identifying selected connection parameters; allocating a dial-in number; notifying participants of the dial-in number; detecting a call from a requesting participant; determining if sufficient resources for the participant are available; and connecting the requesting participant into the reservationless conference. Additionally, the video teleconferencing method allows the conference time to be automatically extended. This is done using timers and reallocating available resources when the original end of conference time arrives. The method also enables the conference to automatically terminate after all conference participants leave the conference.

Abstract: Systems, methods, and devices for improving speaker performance with an acoustic damped port are disclosed. In accordance with various embodiments of the present invention, a damping material is placed around a vented frame of a speaker driver, and substantially covers or fills the vents of the frame. In some embodiments, the damping material results in improved impedance matching with acoustic delay, without required dimensional tuning (as with conventional ports). In some embodiments, it also reduces the air velocity gradients minimizing higher order frequency distortion components. In some embodiments, the damping material also acts as an absorber of energy by coupling to the sound wave. In some embodiments, low-frequency performance is improved in a small-scale design.

Abstract: Systems, methods, and devices for providing high quality speaker performance in a small-scale design are disclosed. In accordance with various embodiments of the present invention, a small-scale speaker assembly is provided which is capable of delivering high-fidelity wideband audio. The speaker assembly includes a speaker driver and a number of electronic components within an enclosure. In some embodiments at least one microphone is included in the assembly. In some embodiments the speaker assembly is well suited for audio conferencing. In some embodiments low-frequency performance is improved in a small-scale design.

Abstract: A videoconferencing unit includes first and second encoder that each encodes a video stream of frames and generates variables indicating changes between frames of the video streams. The unit also includes a controller operatively coupled to the first and second encoders. The controller compares the variables from the encoders and determines first and second bit rates for the first and second encoders based on the comparison. Then, the controller sets the first and second encoders to the first and second bit rates, respectively. Preferably, comparing the variables, determining the bit rates, and setting the bit rates are dynamically repeated as the unit operates. In addition, the dynamic repetition is preferably allowed or limited to one or more predetermined intervals of time.

Abstract: A system, method, and apparatus are disclosed whereby a wireless Personal Area Network such as a Bluetooth piconet may be extended to a remote location beyond the normal range by means of a conferencing connection. The conferencing connection may comprise, for example, one or more ISDN lines or an IP connection between two or more conference endpoints. The broadband connection may include a video channel, an audio channel, a control channel, and a Bluetooth channel.

Abstract: A videoconferencing system includes a first videoconferencing unit and one or more second videoconferencing units, each of which is coupled to one or more networks. The first videoconferencing unit includes a computer network tool for pinging the one or more second videoconferencing units to determine whether the one or more second videoconferencing units are reachable on the network. The one or more second videoconferencing units include a computer network tool for automatically responding to the ping by sending a response to the first videoconferencing unit.

Abstract: A videoconferencing system includes a first videoconferencing unit coupled to a network and associated with a first instant messaging identity. The first videoconferencing unit obtains a second instant messaging identity and automatically sends a request instant message requesting videoconferencing connection information to the second instant messaging identity. A second videoconferencing unit is coupled to the network and is associated with the second instant messaging identity. The second videoconferencing unit receives the request instant message and automatically returns a response instant message including videoconferencing connection information to the first instant messaging identity. The first videoconferencing unit receives the response instant message and automatically obtains the videoconferencing connection information from the response instant message.

Abstract: A videoconferencing system includes a first videoconferencing unit coupled to a network and associated with a first e-mail address. The first videoconferencing unit receives a second e-mail address and automatically sends a request e-mail requesting connection information to the second e-mail address. A second videoconferencing unit is coupled to the network and is associated with the second e-mail address. The second videoconferencing unit receives the request e-mail and automatically returns a response e-mail including connection information to the first e-mail address. The first videoconferencing unit receives the response e-mail and automatically obtains the connection information from the response e-mail. Using the connection information, the first videoconferencing unit initiates a videoconference call with the second videoconference unit.

Abstract: A method for reducing or eliminating coding artifacts in video pictures processed using intra block prediction methods, for example, according to the H.264 standard. The causes of the coding artifacts are identified to be the intra prediction modes with prediction directions that are not in the direction of the raster scan. Filtering the affected blocks with a simple one-dimensional spatial filter will reduce or even eliminate these coding artifacts.

Abstract: Disclosed herein is a middleware server for interfacing a MCU with a communication system and/or a management system within the context of an organization. The middleware server uses modular architecture, making the middleware server highly adaptable to different types of communication systems and management systems. The middleware server is useful for interfacing a MCU with communication devices including POTS telephones, IP-telephones, and multimedia endpoints and with management applications such as exchange servers and databases. The middleware server is useful for managing multimedia sessions according to organizational policies.

Abstract: The present invention provides, in one embodiment, a system and method for concealing video errors. The system encodes, reorders, and packetizes video information into video data packets for transmission over a communication network such that the system conceals errors caused by lost video data packets when the system receives, depacketizes, orders, and decodes the data packets. In one embodiment, the system and method encodes and packetizes video information, such that adjacent macroblocks are not placed in the same video data packets. Additionally, the system and method may provide information accompanying the video data packets to facilitate the decoding process. An advantage to such a scheme is that errors due to video data packet loss are spatially distributed over a video frame. Thus, if regions of data surrounding a lost macroblock are successfully decoded, the decoder may predict motion vectors and spatial content with a higher degree of accuracy, which leads to higher video quality.

Abstract: A system and method for dynamically establishing optimum audio quality in an audio conference is disclosed. A connection with one or more remote communication devices is initially established. An available data rate associated with the connection is then determined. Next, a bandwidth is assigned based on the available data rate. Finally, the assigned bandwidth is adjusted according to the available data rate.