Abstract: A method performed by a processing system includes providing, to an audio service, a virtual microphone-selection corresponding to at least one of a set of audio source devices determined to be in proximity to the processing system and receiving, from the audio service, an output audio stream that is formed from one of a set of source audio streams received from the set of audio source devices and corresponds to the virtual microphone selection.

Abstract: A method for time delay estimation performed by a physical computing system includes passing a first input signal obtained by a first sensor through a filter bank to form a first set of sub-band output signals, passing a second input signal obtained by a second sensor through the filter bank to form a second set of sub-band output signals, the second sensor placed a distance from the first sensor, computing cross-correlation data between the first set of sub-band output signals and the second set of sub-band output signals, and applying a time delay determination function to the cross-correlation to determine a time delay estimation.

Abstract: Various embodiments of the present invention are directed to adaptive methods for reducing acoustic echoes in multichannel audio communication systems. Acoustic echo cancellation methods determine approximate impulse responses characterizing each echo path between loudspeakers and microphones within a room and improve performance based on previously determined impulse responses. In particular, the methods adapt to changes in the room by inferring approximate impulse responses that lie within a model of an impulse response space. Over time the method improves performance by evolving the model into a more accurate space from which to select subsequent approximate impulse responses.

Abstract: A method for time delay estimation performed by a physical computing system includes passing a first input signal obtained by a first sensor through a filter bank to form a first set of sub-band output signals, passing a second input signal obtained by a second sensor through the filter bank to form a second set of sub-band output signals, the second sensor placed a distance from the first sensor, computing cross-correlation data between the first set of sub-band output signals and the second set of sub-band output signals, and applying a time delay determination function to the cross-correlation to determine a time delay estimation.

Abstract: Various embodiments of the present invention are directed to methods for dereverberation of audio generated in a room. In one aspect, a method for dereverberating reverberant digital signals comprises transforming a reverberant digital signal from the time domain into Fourier domain signals using a computing device, each Fourier domain signal corresponding to a subband. For each subband of the Fourier domain signal, the method computes autoregressive model coefficients of the reverberation with the current and previous magnitudes of the Fourier digital signal, and inverse filters the magnitude of the Fourier domain signal using the computing device, based on the autoregressive model coefficients and previous magnitudes of the Fourier digital signal. The method includes inverse transforming the Fourier domain signals with filtered magnitudes into an approximate dereverberated digital signal.

Abstract: Various embodiments of the present invention are directed to methods for dereverberation of audio generated in a room. In one aspect, a method for dereverberating reverberant digital signals comprises transforming a reverberant digital signal from the time domain into Fourier domain signals using a computing device, each Fourier domain signal corresponding to a subband. For each subband of the Fourier domain signal, the method computes autoregressive model coefficients of the reverberation with the current and previous magnitudes of the Fourier digital signal, and inverse filters the magnitude of the Fourier domain signal using the computing device, based on the autoregressive model coefficients and previous magnitudes of the Fourier digital signal. The method includes inverse transforming the Fourier domain signals with filtered magnitudes into an approximate dereverberated digital signal.

Abstract: Various embodiments of the present invention are directed to adaptive methods for reducing acoustic echoes in multichannel audio communication systems. Acoustic echo cancellation methods determine approximate impulse responses characterizing each echo path between loudspeakers and microphones within a room and improve performance based on previously determined impulse responses. In particular, the methods adapt to changes in the room by inferring approximate impulse responses that lie within a model of an impulse response space. Over time the method improves performance by evolving the model into a more accurate space from which to select subsequent approximate impulse responses.

Abstract: Various embodiments of the present invention are directed to a frequency-domain coder/decoder for an audio-conference communication system that includes acoustic-echo-cancellation functionality. In one embodiment of the present invention, an acoustic echo canceller is integrated into the frequency-domain coder/decoder and ameliorates or removes acoustic echoes from audio signals that have been transformed to the frequency domain and divided into subbands by the frequency-domain coder/decoder.

Abstract: The invention features an improved, photo hull based method of generating a three-dimensional representation of a visual scene based upon a set of multi-view reference images. In one aspect, a visual hull containing the visual scene is computed. Visibility of points on the computed visual hull with respect to each reference image is computed. Photo-consistency of points on the computed visual hull along rays back-projected from a desired view is computed based upon reference images having visibility of the points. A photo hull containing the visual scene is generated by stepping along each back-projected ray. The photo-consistent points visible in the desired view are colored using color information contained in reference images having visibility of the photo-consistent points.

Abstract: A system and method for enabling efficient error correction and encryption using wavelet transforms over finite fields. The system and method utilizes the combination of a channel encoder and channel decoder to correct errors to source data after transmission over a physical channel or storage in a storage medium. The channel encoder mathematically generates a set of wavelet coefficients by performing a combination of filtering and/or processing of a received message vector. The wavelet coefficients are then utilized by the channel encoder to cause its filters to transform message data into transmission data. The channel decoder receives the transmitted source data in the form of a code word/channel error combination and performs filtering to render a syndrome, representative of the channel error. Analysis of the syndrome is performed to determine the actual error, which is utilized to derive the actual source data.

Abstract: A novel method for the measurement of the Thiele-Small small-signal parameters of a moving-coil electromagnetic transducer driver is disclosed. The technique is based on a time domain analysis of the transient response of a loudspeaker voice coil circuit to a current step of excitation. By sampling the damped sinusoidal transient generated by such an excitation, the loud-speaker parameters can be calculated from a linear predictive analysis of the recorded data.