Abstract: A method of updating the state of a decoder that decodes a wide-band signal including a plurality of sub-band signals, including: receiving the plurality of sub-band signals; for each sub-band signal, storing portions of that sub-band signal in a respective buffer; responsive to determining that a portion of the wide-band signal is degraded, performing a packet loss concealment algorithm to determine wide-band replacement data for the degraded portion; selecting a portion of the sub-band signal stored in each buffer in dependence on the determined wide-band replacement data; and updating the state of the decoder using the selected portions.

Abstract: The present invention relates to adaptive beamforming in audio systems. More specifically, aspects of the invention relate to a method for adaptively estimating a target sound signal by establishing a simulation model simulating an audio environment comprising: a plurality of spatially separated microphones, a target sound source, and a number of audio noise sources.

Abstract: The present invention relates to adaptive beamforming in audio systems. More specifically, aspects of the invention relate to a method for adaptively estimating a target sound signal by establishing a simulation model simulating an audio environment comprising: a plurality of spatially separated microphones, a target sound source, and a number of audio noise sources.

Abstract: A method, apparatus, and manufacture for beamforming is provided. Parameters based on sets of pre-determined beamforming weights are stored. Each of the sets of pre-determined beamforming weights has a corresponding integral index number. Each input microphone signal is transformed to the frequency domain to provide a corresponding transformed signal. Each of the transformed signals includes a plurality of subbands. Next, an index number is determined representing an optimal set of beamforming weights for the transformed signals. Then, a set of beamforming weights is applied to each subband of each of the transformed signals to provide a weighted signal. The set corresponds to the determined index number. A time domain signal is then provided by combining each of the weighted signals.

Abstract: The present invention relates to adaptive beamforming in audio systems. More specifically, aspects of the invention relate to a method for adaptively estimating a target sound signal by establishing a simulation model simulating an audio environment comprising: a plurality of spatially separated microphones, a target sound source, and a number of audio noise sources.

Abstract: A method of updating the state of a decoder that decodes a wide-band signal including a plurality of sub-band signals, including: receiving the plurality of sub-band signals; for each sub-band signal, storing portions of that sub-band signal in a respective buffer; responsive to determining that a portion of the wide-band signal is degraded, performing a packet loss concealment algorithm to determine wide-band replacement data for the degraded portion; selecting a portion of the sub-band signal stored in each buffer in dependence on the determined wide-band replacement data; and updating the state of the decoder using the selected portions.

Abstract: A method and system for clear signal capture comprehend several individual aspects that address specific problems in improved ways. In addition, the method and system also comprehend a hands-free implementation that is a practical solution to a very complex problem. Individual aspects comprehended related to echo and noise reduction, and divergence control.

Abstract: A method and system for clear signal capture comprehend several individual aspects that address specific problems in improved ways. In addition, the method and system also comprehend a hands-free implementation that is a practical solution to a very complex problem. Individual aspects comprehended related to echo and noise reduction, and divergence control.

Abstract: A method and system for clear signal capture comprehend several individual aspects that address specific problems in improved ways. In addition, the method and system also comprehend a hands-free implementation that is a practical solution to a very complex problem. Individual aspects comprehended related to echo and noise reduction, and divergence control.

Abstract: A method and system for clear signal capture comprehend several individual aspects that address specific problems in improved ways. In addition, the method and system also comprehend a hands-free implementation that is a practical solution to a very complex problem. Individual aspects comprehended related to echo and noise reduction, and divergence control.

Abstract: Improved vocal signals are obtained in headsets and similar devices by including a microphone inside a chamber formed at least in part by the wearer's ear. This second microphone provides a reduced noise input signal. The reduced noise signal is corrected by input from another microphone, located outside the chamber. This correction can include echo cancellation, spectral shaping, frequency extension, and the like.

Abstract: A method and system for clear signal capture comprehend several individual aspects that address specific problems in improved ways. In addition, the method and system also comprehend a hands-free implementation that is a practical solution to a very complex problem. Individual aspects comprehended related to echo and noise reduction, and divergence control.

Abstract: A method and system for clear signal capture comprehend several individual aspects that address specific problems in improved ways. In addition, the method and system also comprehend a hands-free implementation that is a practical solution to a very complex problem. Individual aspects comprehended related to echo and noise reduction, and divergence control.

Abstract: A method and system for clear signal capture comprehend several individual aspects that address specific problems in improved ways. In addition, the method and system also comprehend a hands-free implementation that is a practical solution to a very complex problem. Individual aspects comprehended related to echo and noise reduction, and divergence control.

Abstract: A method and system for clear signal capture comprehend several individual aspects that address specific problems in improved ways. In addition, the method and system also comprehend a hands-free implementation that is a practical solution to a very complex problem. Individual aspects comprehended related to echo and noise reduction, and divergence control.

Abstract: A method and system for clear signal capture comprehend several individual aspects that address specific problems in improved ways. In addition, the method and system also comprehend a hands-free implementation that is a practical solution to a very complex problem. Individual aspects comprehended related to echo and noise reduction, and divergence control.

Abstract: A method and system for clear signal capture comprehend several individual aspects that address specific problems in improved ways. In addition, the method and system also comprehend a hands-free implementation that is a practical solution to a very complex problem.

Abstract: A method of reducing noise by cascading a plurality of noise reduction algorithms is provided. A sequence of noise reduction algorithms are applied to the noisy signal. The noise reduction algorithms are cascaded together, with the final noise reduction algorithm in the sequence providing the system output signal. The sequence of noise reduction algorithms includes a plurality of noise reduction algorithms that are sufficiently different from each other such that resulting distortions and artifacts are sufficiently different to result in reduced human perception of the artifact and distortion levels in the system output signal.

Abstract: Improved vocal signals are obtained in headsets and similar devices by including a microphone inside a chamber formed at least in part by the wearer's ear. This second microphone provides a reduced noise input signal. The reduced noise signal is corrected by input from another microphone, located outside the chamber. This correction can include echo cancellation, spectral shaping, frequency extension, and the like.

Abstract: The presence of speech in a filtered speech signal is detected for the purpose of suspending noise level calculations during periods of speech. A received speech signal is split into a plurality of subband signals. A subband variable gain is determined for each subband based on an estimation of the noise level in the received voice signal and on an envelope of the received signal in each subband. Each subband signal is multiplied by the subband variable gain for that subband. The subband signals are combined to produce an output voice signal.