Abstract: A method for estimating an audio device location in an environment may involve obtaining direction of arrival (DOA) data for each audio device of a plurality of audio devices in the environment and determining interior angles for each of a plurality of triangles based on the DOA data. Each triangle may have vertices that correspond with audio device locations. The method may involve determining a side length for each side of each of the triangles, performing a forward alignment process of aligning each of the plurality of triangles produce a forward alignment matrix and performing a reverse alignment process of aligning each of the plurality of triangles in a reverse sequence to produce a reverse alignment matrix. A final estimate of each audio device location may be based, at least in part, on values of the forward alignment matrix and values of the reverse alignment matrix.

Abstract: In an audio conferencing environment, including multiple users participating by means of a series of associated audio input devices for the provision of audio input, and a series of audio output devices for the output of audio output streams to the multiple users, with the audio input and output devices being interconnected to a mixing control server for the control and mixing of the audio inputs from each audio input devices to present a series of audio streams to the audio output devices, a method of reducing the effects of cross talk pickup of at least a first audio conversation by multiple audio input devices, the method including the steps of: (a) monitoring the series of audio input devices for the presence of a duplicate audio conversation input from at least two input audio sources in an audio output stream; and (b) where a duplicate audio conversation input is detected, suppressing the presence of the duplicate audio conversation input in the audio output stream.

Abstract: In an audio conferencing environment, including multiple users participating by means of a series of associated audio input devices for the provision of audio input, and a series of audio output devices for the output of audio output streams to the multiple users, with the audio input and output devices being interconnected to a mixing control server for the control and mixing of the audio inputs from each audio input devices to present a series of audio streams to the audio output devices, a method of reducing the effects of cross talk pickup of at least a first audio conversation by multiple audio input devices, the method including the steps of: (a) monitoring the series of audio input devices for the presence of a duplicate audio conversation input from at least two input audio sources in an audio output stream; and (b) where a duplicate audio conversation input is detected, suppressing the presence of the duplicate audio conversation input in the audio output stream.

Abstract: In some embodiments, a method for selecting at least one layer of a spatially layered, encoded audio signal. Typical embodiments are teleconferencing methods in which at least one of a set of nodes (endpoints, each of which is a telephone system, and optionally also a server) is configured to perform audio coding in response to soundfield audio data to generate spatially layered encoded audio including any of a number of different subsets of a set of layers, the set of layers including at least one monophonic layer, at least one soundfield layer, and optionally also at least one metadata layer comprising metadata indicative of at least one processing operation to be performed on the encoded audio. Other aspects are systems configured (e.g., programmed) to perform any embodiment of the method, and computer readable media which store code for implementing any embodiment of the method or steps thereof.

Abstract: Embodiments of client device and method for audio or video conferencing are described. An embodiment includes an offset detecting unit, a configuring unit, an estimator and an output unit. The offset detecting unit detects an offset of speech input to the client device. The configuring unit determines a voice latency from the client device to every far end. The estimator estimates a time when a user at the far end perceives the offset based on the voice latency. The output unit outputs a perceivable signal indicating that a user at the far end perceives the offset based on the time estimated for the far end. The perceivable signal is helpful to avoid collision between parties.

Abstract: The invention provides a layered audio coding format with a monophonic layer and at least one sound field layer. A plurality of audio signals is decomposed, in accordance with decomposition parameters controlling the quantitative properties of an orthogonal energy-compacting transform, into rotated audio signals. Further, a time-variable gain profile specifying constructively how the rotated audio signals may be processed to attenuate undesired audio content is derived. The monophonic layer may comprise one of the rotated signals and the gain profile. The sound field layer may comprise the rotated signals and the decomposition parameters. In one embodiment, the gain profile comprises a cleaning gain profile with the main purpose of eliminating non-speech components and/or noise. The gain profile may also comprise mutually independent broadband gains.

Abstract: The invention provides a layered audio coding format with a monophonic layer and at least one sound field layer. A plurality of audio signals is decomposed, in accordance with decomposition parameters controlling the quantitative properties of an orthogonal energy-compacting transform, into rotated audio signals. Further, a time-variable gain profile specifying constructively how the rotated audio signals may be processed to attenuate undesired audio content is derived. The monophonic layer may comprise one of the rotated signals and the gain profile. The sound field layer may comprise the rotated signals and the decomposition parameters. In one embodiment, the gain profile comprises a cleaning gain profile with the main purpose of eliminating non-speech components and/or noise. The gain profile may also comprise mutually independent broadband gains.

Abstract: Method for measuring level of speech determined by an audio signal in a manner which corrects for and reduces the effect of modification of the signal by the addition of noise thereto and/or amplitude compression thereof, and a system configured to perform any embodiment of the method. In some embodiments, the method includes steps of generating frequency banded, frequency-domain data indicative of an input speech signal, determining from the data a Gaussian parametric spectral model of the speech signal, and determining from the parametric spectral model an estimated mean speech level and a standard deviation value for each frequency band of the data; and generating speech level data indicative of a bias corrected mean speech level for each frequency band, including using at least one correction value to correct the estimated mean speech level for the frequency band, where each correction value has been predetermined using a reference speech model.

Abstract: An audio processing method and an audio processing apparatus are described. A mono-channel audio signal is transformed into a plurality of first subband signals. Proportions of a desired component and a noise component are estimated in each of the subband signals. Second subband signals corresponding respectively to a plurality of channels are generated from each of the first subband signals. Each of the second subband signals comprises a first component and a second component obtained by assigning a spatial hearing property and a perceptual hearing property different from the spatial hearing property to the desired component and the noise component in the corresponding first subband signal respectively, based on a multi-dimensional auditory presentation method. The second subband signals are transformed into signals for rendering with the multi-dimensional auditory presentation method.

Abstract: The invention provides a layered audio coding format with a monophonic layer and at least one sound field layer. A plurality of audio signals is decomposed, in accordance with decomposition parameters controlling the quantitative properties of an orthogonal energy-compacting transform, into rotated audio signals. Further, a time-variable gain profile specifying constructively how the rotated audio signals may be processed to attenuate undesired audio content is derived. The monophonic layer may comprise one of the rotated signals and the gain profile. The sound field layer may comprise the rotated signals and the decomposition parameters. In one embodiment, the gain profile comprises a cleaning gain profile with the main purpose of eliminating non-speech components and/or noise. The gain profile may also comprise mutually independent broadband gains.

Abstract: The invention provides a layered audio coding format with a monophonic layer and at least one sound field layer. A plurality of audio signals is decomposed, in accordance with decomposition parameters controlling the quantitative properties of an orthogonal energy-compacting transform, into rotated audio signals. Further, a time-variable gain profile specifying constructively how the rotated audio signals may be processed to attenuate undesired audio content is derived. The monophonic layer may comprise one of the rotated signals and the gain profile. The sound field layer may comprise the rotated signals and the decomposition parameters. In one embodiment, the gain profile comprises a cleaning gain profile with the main purpose of eliminating non-speech components and/or noise. The gain profile may also comprise mutually independent broadband gains.

Abstract: In some embodiments, a method for selecting at least one layer of a spatially layered, encoded audio signal. Typical embodiments are teleconferencing methods in which at least one of a set of nodes (endpoints, each of which is a telephone system, and optionally also a server) is configured to perform audio coding in response to soundfield audio data to generate spatially layered encoded audio including any of a number of different subsets of a set of layers, the set of layers including at least one monophonic layer, at least one soundfield layer, and optionally also at least one metadata layer comprising metadata indicative of at least one processing operation to be performed on the encoded audio. Other aspects are systems configured (e.g., programmed) to perform any embodiment of the method, and computer readable media which store code for implementing any embodiment of the method or steps thereof.

Abstract: An audio processing method and an audio processing apparatus are described. A mono-channel audio signal is transformed into a plurality of first subband signals. Proportions of a desired component and a noise component are estimated in each of the subband signals. Second subband signals corresponding respectively to a plurality of channels are generated from each of the first subband signals. Each of the second subband signals comprises a first component and a second component obtained by assigning a spatial hearing property and a perceptual hearing property different from the spatial hearing property to the desired component and the noise component in the corresponding first subband signal respectively, based on a multi-dimensional auditory presentation method. The second subband signals are transformed into signals for rendering with the multi-dimensional auditory presentation method.

Abstract: Method for measuring level of speech determined by an audio signal in a manner which corrects for and reduces the effect of modification of the signal by the addition of noise thereto and/or amplitude compression thereof, and a system configured to perform any embodiment of the method. In some embodiments, the method includes steps of generating frequency banded, frequency-domain data indicative of an input speech signal, determining from the data a Gaussian parametric spectral model of the speech signal, and determining from the parametric spectral model an estimated mean speech level and a standard deviation value for each frequency band of the data; and generating speech level data indicative of a bias corrected mean speech level for each frequency band, including using at least one correction value to correct the estimated mean speech level for the frequency band, where each correction value has been predetermined using a reference speech model.

Abstract: In a class of embodiments, a method and system for characterizing a display device using a camera device (e.g., a consumer grade, mobile camera device) to sense light emitted by the display device during display of one or more test images, and to generate measurement data characterizing the display and/or calibration data useful for calibrating the display device. In some embodiments, the display device is adjusted or calibrated manually in response to such calibration data, for example by user actuation of display device controls in response to calibration data indicative of display setting recommendations. In some embodiments, the test image(s) are embedded in a video stream (e.g., overlaid on arbitrary video content). For example, test images including appropriate timing information and calibration sequences are embedded in a video stream for display by the display device. In some embodiments, the user is prompted to align the camera device (e.g.

Abstract: In a class of embodiments, a method and system for characterizing a display device using a camera device (e.g., a consumer grade, mobile camera device) to sense light emitted by the display device during display of one or more test images, and to generate measurement data characterizing the display and/or calibration data useful for calibrating the display device. In some embodiments, the display device is adjusted or calibrated manually in response to such calibration data, for example by user actuation of display device controls in response to calibration data indicative of display setting recommendations. In some embodiments, the test image(s) are embedded in a video stream (e.g., overlaid on arbitrary video content). For example, test images including appropriate timing information and calibration sequences are embedded in a video stream for display by the display device. In some embodiments, the user is prompted to align the camera device (e.g.