A novel rank reduction scheme is introduced for adaptive filtering problems. This rank reduction method uses a cross-spectral metric to select the optimal lower dimensional subspace for reduced-rank adaptive filtering as a function of the basis vectors of the full-rank space.

Reduced-rank adaptive filtering

Systems and methods for providing single microphone noise suppression fallback are provided. In exemplary embodiments, primary and secondary acoustic signals are received. A single microphone noise estimate may be generated based on the primary acoustic signal, while a dual microphone noise estimate may be generated based on the primary and secondary acoustic signals. A combined noise estimate based on the single and dual microphone noise estimates is then determined. Using the combined noise estimate, a gain mask may be generated and applied to the primary acoustic signal to generate a noise suppressed signal. Subsequently, the noise suppressed signal may be output.

System and method for providing single microphone noise suppression fallback

Systems and methods for utilizing inter-microphone level differences (ILD) to attenuate noise and enhance speech are provided. In exemplary embodiments, primary and secondary acoustic signals are received by omni-directional microphones, and converted into primary and secondary electric signals. A differential microphone array module processes the electric signals to determine a cardioid primary signal and a cardioid secondary signal. The cardioid signals are filtered through a frequency analysis module which takes the signals and mimics a cochlea implementation (i.e., cochlear domain). Energy levels of the signals are then computed, and the results are processed by an ILD module using a non-linear combination to obtain the ILD. In exemplary embodiments, the non-linear combination comprises dividing the energy level associated with the primary microphone by the energy level associated with the secondary microphone. The ILD is utilized by a noise reduction system to enhance the speech of the primary acoustic signal.

System and method for utilizing omni-directional microphones for speech enhancement

A method for estimating mean opinion score or naturalness of synthesized speech is provided. The method includes using an objective measure that has components derived directly from textual information used to form synthesized utterances. The objective measure has a high correlation with mean opinion score such that a relationship can be formed between the objective measure and corresponding mean opinion score. An estimated mean opinion score can be obtained easily from the relationship when the objective measure is applied to utterances of a modified speech synthesizer.

Objective measure for estimating mean opinion score of synthesized speech

A method is provided for optimizing an objective measure used to estimate mean opinion score or naturalness of synthesized speech from a speech synthesizer. The method includes using an objective measure that has components derived directly from textual information used to form synthesized utterances. The objective measure has a high correlation with mean opinion score such that a relationship can be formed between the objective measure and corresponding mean opinion score. The objective measure is altered to provide a different function of textual information derived from the utterances so as to improve the relationship between the scores of the objective measure and subjective ratings of the synthesized utterances.

Optimization of an objective measure for estimating mean opinion score of synthesized speech

A method and system for adaptively filtering a speech signal. The method includes decomposing the signal into subbands, which may include performing a discrete Fourier transform on the signal to provide approximately orthogonal components. The method also includes determining a speech quality indicator for each subband, which may include estimating a signal-to-noise ratio for each subband. The method also includes selecting a filter for filtering each subband depending on the speech quality indicator, which may include estimating parameters for the filter based on a clean speech signal. The method further includes determining an overall average error for the filtered subbands, which may include calculating a mean-squared error. The method still further includes identifying at least one filtered subband which, if excluded from the filtered speech signal, would reduce the overall average error determined, and combining, with exception of the filtered subbands identified, the filtered subbands to provide an estimated filtered speech signal. The system includes filters and software for performing the method.

Method and system for adaptive filtering of speech signals using signal-to-noise ratio to choose subband filter bank

A method and system for objectively evaluating the quality of speech in a voice communication system A plurality of speech reference vectors is first obtained based on a plurality of clean speech samples A corrupted speech signal is received and processed to determine a plurality of distortions derived from a plurality of distortion measures based on the plurality of speech reference vectors The plurality of distortions are processed by a non-linear neural network model to generate a subjective score representing user acceptance of the corrupted speech signal The non-linear neural network model is first trained on clean speech samples as well as corrupted speech samples through the use of backpropagation to obtain the weights and bias terms necessary to predict subjective scores from several objective measures

Method and system for objectively evaluating speech

A speech signal divided into frames, each frame having a sound type, and a class is determined for each frame depending on the sound type of the frame. One of multiple filters is selected for each frame depending on the class of the frame. Each frame is filtered according to the filter selected, and the filtered frames combined to provide a filtered speech signal. The system includes filters and software.

Method and system for region based filtering of speech

A method is disclosed for identifying corrupted speech signals in a call receiving mode of a voice messaging system. The method includes the step of receiving a message signal. The message signal represents an audio message. The method next includes the step of determining a signal quality. The signal quality is then compared to a threshold. If the signal quality is at least as great as the threshold, the audio data representing the message signal is stored in a memory. If the signal quality is not as great as the threshold, an indication signal is transmitted indicating that the signal quality is poor. A system is also disclosed for implementing the steps of the method.

Marvin L. Vis

Papers

Method and system for adaptive filtering of speech signals using signal-to-noise ratio to choose subband filter bank

Method and system for objectively evaluating speech

Method and system for region based filtering of speech

Method and system for identifying a corrupted speech message signal