Abstract: An acoustic-scene interpretation apparatus can have a transducer configured to convert an acoustic signal to a corresponding electrical signal. A feature extractor can receive a sequence of frames representing the electrical signal and extract a plurality of acoustic features corresponding to each frame. An acoustic-scene classifier can be configured to determine a most-likely acoustic state for each frame in the sequence of frames in correspondence with the respective plurality of acoustic features corresponding to the frame and a selected probability distribution of duration of an acoustic state for each of one or more classes of acoustic scenes. Each respective probability distribution of duration can correspond to a selected class of acoustic scenes. The correspondence between acoustic state and probability distribution of duration can be learned from training data corresponding to each of a plurality of classes of acoustic scenes. Related methods also are disclosed.

Abstract: An acoustic-scene interpretation apparatus can have a transducer configured to convert an acoustic signal to a corresponding electrical signal. A feature extractor can receive a sequence of frames representing the electrical signal and extract a plurality of acoustic features corresponding to each frame. An acoustic-scene classifier can be configured to determine a most-likely acoustic state for each frame in the sequence of frames in correspondence with the respective plurality of acoustic features corresponding to the frame and a selected probability distribution of duration of an acoustic state for each of one or more classes of acoustic scenes. Each respective probability distribution of duration can correspond to a selected class of acoustic scenes. The correspondence between acoustic state and probability distribution of duration can be learned from training data corresponding to each of a plurality of classes of acoustic scenes. Related methods also are disclosed.

Abstract: A method and system for optimally repairing a clipped audio signal. Clipping occurs when a waveform exceeds a dynamic range of a recording device. Portions of an audio signal exceeding the dynamic range or saturation level of the recording device are clipped, causing distortion when the clipped recorded signal is played. To address this problem, successive frames of the clipped audio data are repaired to fill in gaps where the data were clipped. For each frame, an iterative process repetitively estimates an auto-covariance and detects clipped samples in the frame or a sub-frame in order to compute a least-squares solution for the frame that interpolates the clipped data. The process can cause inverted peaks in the repaired data, which must then be rectified to produced corrected repaired data. The corrected repaired data for the successive frames are recombined using interpolation, to produce a complete repaired audio data set.

Abstract: The restoration of melody perception is a key remaining challenge in cochlear implants. A novel sound coding strategy is proposed that converts an input audio signal into time-varying electrically stimulating pulse trains. A sound is first split into several frequency sub-bands with a fixed filter bank or a dynamic filter bank tracking harmonics in sounds. Each sub-band signal is coherently downward shifted to a low-frequency base band. These resulting coherent envelope signals have Hermitian symmetric frequency spectrums and are thus real-valued. A peak detector or high-rate sampler of half-wave rectified coherent envelope signals in each sub-band further converts the coherent envelopes into rate-varying, interleaved pulse trains. Acoustic simulations of cochlear implants using this new technique with normal hearing listeners, showed significant improvement in melody recognition over the most common conventional stimulation approach used in cochlear implants.

Abstract: The restoration of melody perception is a key remaining challenge in cochlear implants. A novel sound coding strategy is proposed that converts an input audio signal into time-varying electrically stimulating pulse trains. A sound is first split into several frequency sub-bands with a fixed filter bank or a dynamic filter bank tracking harmonics in sounds. Each sub-band signal is coherently downward shifted to a low-frequency base band. These resulting coherent envelope signals have Hermitian symmetric frequency spectrums and are thus real-valued. A peak detector or high-rate sampler of half-wave rectified coherent envelope signals in each sub-band further converts the coherent envelopes into rate-varying, interleaved pulse trains. Acoustic simulations of cochlear implants using this new technique with normal hearing listeners, showed significant improvement in melody recognition over the most common conventional stimulation approach used in cochlear implants.

Abstract: A method and system for optimally repairing a clipped audio signal. Clipping occurs when a waveform exceeds a dynamic range of a recording device. Portions of an audio signal exceeding the dynamic range or saturation level of the recording device are clipped, causing distortion when the clipped recorded signal is played. To address this problem, successive frames of the clipped audio data are repaired to fill in gaps where the data were clipped. For each frame, an iterative process repetitively estimates an auto-covariance and detects clipped samples in the frame or a sub-frame in order to compute a least-squares solution for the frame that interpolates the clipped data. The process can cause inverted peaks in the repaired data, which must then be rectified to produced corrected repaired data. The corrected repaired data for the successive frames are recombined using interpolation, to produce a complete repaired audio data set.

Abstract: Information is estimated to fill-in even relatively long gaps (e.g., up to 250 ms) that occur in a signal due to physical errors in media or transmission, where the omitted information causes signal distortion. The signal is first divided into a plurality of subbands, since the gaps in each subband are individually easier to interpolate. Coherent demodulation is then employed on each subband signal to reduce the time-varying signals to a collection of pairs of frequency-modulated carriers multiplied by complex-valued envelopes, or modulators. Standard interpolation is then separately applied to the modulators and carriers of these pairs to fill-in the gaps in each of the subbands, and the interpolated pairs are remodulated. The resulting interpolated signals from each of the subbands are recombined to form the final interpolated output signal in which the gaps are filled in with estimated data.