Abstract: A video teleconferencing directional microphone includes two microphone elements arranged coincidentally on a vertical axis. The two microphone elements are placed on a supporting surface so that a first microphone element is on the surface, and the second microphone elements are elevated above the supporting surface. The directional microphone also includes filters, an adder assembly, and an equalizer, which are used to shape the directivity pattern of the directional microphone into a toroid sensitivity pattern. The toroid sensitivity pattern increases sensitivity in the direction of a sound source of interest, while simultaneously reduces sensitivity to any sound waves generated by noise sources from certain elevation angles.

Abstract: An arrangement and a method of localizing active speakers in a video conference includes a localization device that locates at least one microphone relative to the camera, while the at least one microphone in turn localizes the relative positions of audio sources. As the microphones are usually positioned close to the audio source in a video conference, the ratio of the distance between microphones relative to the distance between table microphone and audio source is reduced. Thus, the microphones are able to determine the positions of the audio sources with a higher resolution than if placed close to the camera. When the respective positions of the microphones relative to the camera and the audio source are known, the position of the audio source relative to the camera is then determined by means of geometrical calculations.

Abstract: Different sampling rates between a playout unit and a capture unit are compensated for via a system, method and computer program product. The playout unit receives samples from a computational unit, and the capture unit sends samples to the computational unit. A playout FIFO buffer operates in a playout time domain, and a capture FIFO buffer operates in a capture time domain. The computational unit is synchronized to a common clock. A first relationship is calculated between the common clock and a playout fifo buffer read pointer, and a second relationship is calculated between the common clock and a capture FIFO buffer write pointer. For each sample in the playout time domain a corresponding sample in the samples from said computational unit is found and sent to the playout FIFO buffer. For each sample in the common clock time domain the corresponding sample in the capture time domain is found and sent to the computational unit.

Abstract: An audio communication system and method with improved acoustic characteristics features a stereo detector that is introduced in the echo canceller of the system. When stereo in far-end audio is detected, converging of the adaptive mono model of the canceller is suspended, and when stereo in far-end audio is not detected, converging of the adaptive stereo model of the canceller is suspended. The system may also be extended with a miscellaneous processing unit configured to attenuate the signal at certain events implying a large stereo echo contribution. A stereo collapsing unit is also introduced on the channels of the far-end audio to remove the stereo image at certain events to further suppress the echo contribution.

Abstract: A sound recording and/or sound reproducing part loosely anchored/connected to a second part wherein the sound recording and/or sound reproducing part is levitated on a magnetic field. The sound recording and/or sound reproducing part and the second part in combination forms a first sound recording and/or sound reproducing device. The sound recording and/or sound reproducing part can be used within on or more of the following devices; a loud speaking telephone, a telephone, wireless communication equipment, a mobile phone, a Bluetooth device, a videoconference terminal, a hearing aid or active ear protection equipment.

Abstract: An audio source tracking arrangement, integrated in or connected to a video conference system, for determining a position of a source creating a sound, including: at least an audio signal processing module configured to determine the position of the source creating the sound based on a plurality of audio signals originating from the source respectively captured by a plurality of microphones; and one or more microphone housings, respectively encapsulating at least one of the plurality of microphones, the one or more microphone housings including a cavity in which at least one of the plurality of microphones is localized, an aperture on a surface of the microphone housing, and a channel extending from the cavity to the aperture, wherein the channel and the cavity are dimensioned to form an acoustical amplifier with a frequency response having one or more high frequency peaks in a frequency band of the sound.

Abstract: An arrangement and a method of a videoconferencing end-point provide for eye contact between users at different sites. A conference camera is placed in the light beam of a projector apparatus without degrading the camera-captured view by the projector light. This is provided by generating a black spot in the light beam covering the camera lens, and by adjusting the position and/or size of the black spot according to a camera position detector, which may be the camera itself or one or more light sensors around and close to the camera lens.

Abstract: The present invention relates to an audio communication system and method with improved acoustic characteristics. The present invention utilizes the Haas effect to simulate a sound picture that gives a better subjective impression of stereo sound compared to the objective stereo image. It provides a system and a method for presenting the stereo image in an optimal way for an associated echo canceller without compromising with the subjective perception of stereo.

Abstract: An echo canceller processing echo, noise and near end talk in a narrower, but still intelligible, frequency band for reducing required processing power and complexity. In a preferred embodiment of the present invention, an input audio signal of captured sound in an audio communication system is decimated and then divided into a number of sub bands by an analyze filter. Each sub band is processed as in background audio echo cancelling by subtracting the signal with an echo estimate from a model of the acoustic signal in the respective sub band, except from that the signal is also bypassed, adjusted by a filter and subtracted from the processed signal. The resulting signals are then recombined by a synthesize filter and interpolated to the original sampling rate and bandwidth. Finally, the output from the synthesize filter is added to the input audio signal, which has been delayed and adjusted by a filter.

Abstract: An arrangement and a method of a videoconferencing end-point provide for eye contact between users at different sites. A conference camera is placed in the light beam of a projector apparatus without degrading the camera-captured view by the projector light. This is provided by generating a black spot in the light beam covering the camera lens, and by adjusting the position and/or size of the black spot according to a camera position detector, which may be the camera itself or one or more light sensors around and close to the camera lens.

Abstract: An echo canceller processing echo, noise and near end talk in a narrower, but still intelligible, frequency band for reducing required processing power and complexity. In a preferred embodiment of the present invention, an input audio signal of captured sound in an audio communication system is decimated and then divided into a number of sub bands by an analyze filter. Each sub band is processed as in background audio echo cancelling by subtracting the signal with an echo estimate from a model of the acoustic signal in the respective sub band, except from that the signal is also bypassed, adjusted by a filter and subtracted from the processed signal. The resulting signals are then recombined by a synthesize filter and interpolated to the original sampling rate and bandwidth. Finally, the output from the synthesize filter is added to the input audio signal, which has been delayed and adjusted by a filter.