Abstract: The present technology provides adaptive noise reduction of an acoustic signal using a sophisticated level of control to balance the tradeoff between speech loss distortion and noise reduction. The energy level of a noise component in a sub-band signal of the acoustic signal is reduced based on an estimated signal-to-noise ratio of the sub-band signal, and further on an estimated threshold level of speech distortion in the sub-band signal. In embodiments, the energy level of the noise component in the sub-band signal may be reduced to no less than a residual noise target level. Such a target level may be defined as a level at which the noise component ceases to be perceptible.

Abstract: The present technology provides a robust noise suppression system that may concurrently reduce noise and echo components in an acoustic signal while limiting the level of speech distortion. An acoustic signal may be received and transformed to cochlear domain sub-band signals. Features, such as pitch, may be identified and tracked within the sub-band signals. Initial speech and noise models may be then be estimated at least in part from a probability analysis based on the tracked pitch sources. Speech and noise models may be resolved from the initial speech and noise models and noise reduction may be performed on the sub-band signals. An acoustic signal may be reconstructed from the noise-reduced sub-band signals.

Abstract: Systems and methods for utilizing inter-microphone level differences to attenuate noise and enhance speech are provided. In exemplary embodiments, energy estimates of acoustic signals received by a primary microphone and a secondary microphone are determined in order to determine an inter-microphone level difference (ILD). This ILD in combination with a noise estimate based only on a primary microphone acoustic signal allow a filter estimate to be derived. In some embodiments, the derived filter estimate may be smoothed. The filter estimate is then applied to the acoustic signal from the primary microphone to generate a speech estimate.

Abstract: Systems and methods for utilizing inter-microphone level differences to attenuate noise and enhance speech are provided. In exemplary embodiments, energy estimates of acoustic signals received by a primary microphone and a secondary microphone are determined in order to determine an inter-microphone level difference (ILD). This ILD in combination with a noise estimate based only on a primary microphone acoustic signal allow a filter estimate to be derived. In some embodiments, the derived filter estimate may be smoothed. The filter estimate is then applied to the acoustic signal from the primary microphone to generate a speech estimate.

Abstract: A system and method are disclosed for transient detection and modification in audio signals. Digital signal processing techniques are used to detect transients and modify an audio signal to enhance or suppress such transients, as desired. A transient audio event is detected in a first portion of the audio signal. A graded response to the detected transient audio event is determined. The first portion of the audio signal is modified in accordance with the graded response. The extent of enhancement or suppression (as applicable) may be determined at least in part by a measure of the significance or magnitude of the transient.

Abstract: Systems and methods for providing acoustic echo cancellation are provided. Primary and secondary acoustic signals are received by a communication device. The acoustic signals may include loudspeaker leakage. A null coefficient is then adaptively determined for each subband of the secondary acoustic signal. The null coefficient is applied to the secondary acoustic signal to generate a coefficient-modified signal. The coefficient-modified signal is subtracted from the primary acoustic signal to generate a masked acoustic signal with reduced or no echo. The masked acoustic signal may be output.

Abstract: The present technology provides adaptive noise reduction of an acoustic signal using a sophisticated level of control to balance the tradeoff between speech loss distortion and noise reduction. The energy level of a noise component in a sub-band signal of the acoustic signal is reduced based on an estimated signal-to-noise ratio of the sub-band signal, and further on an estimated threshold level of speech distortion in the sub-band signal. In embodiments, the energy level of the noise component in the sub-band signal may be reduced to no less than a residual noise target level. Such a target level may be defined as a level at which the noise component ceases to be perceptible.

Abstract: Systems and methods for adaptive processing of a close microphone array in a noise suppression system are provided. A primary acoustic signal and a secondary acoustic signal are received. In exemplary embodiments, a frequency analysis is performed on the acoustic signals to obtain frequency sub-band signals. An adaptive equalization coefficient may then be applied to a sub-band signal of the secondary acoustic signal. A forward-facing cardioid pattern and a backward-facing cardioid pattern are then generated based on the sub-band signals. Utilizing cardioid signals of the forward-facing cardioid pattern and backward-facing cardioid pattern, noise suppression may be performed. A resulting noise suppressed signal is output.

Abstract: 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.

Abstract: 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.

Abstract: Systems and methods for providing voice equalization are provided. In exemplary embodiments, acoustic signals are received from both a near-end and a far-end environment. A power spectrum estimate for the far-end signal and a noise estimate based on the near-end signal are determined. A voice equalization mask based on the power spectrum estimate of the far-end signal and the noise estimate based on the near-end signal is generated and applied to the far-end signal to obtain a modified signal. The modified signal may then be dynamically processed to obtain an enhanced audio signal. The enhanced audio signal is then output.

Abstract: Systems and methods for adaptively classifying audio sources are provided. In exemplary embodiments, at least one acoustic signal is received. One or more acoustic features based on the at least one acoustic signal are derived. A global summary of acoustic features based, at least in part, on the derived one or more acoustic features is determined. Further, an instantaneous global classification based on a global running estimate and the global summary of acoustic features is determined. The global running estimates may be updated and an instantaneous local classification based, at least in part, on the one or more acoustic features may be derived. One or more spectral energy classifications based, at least in part, on the instantaneous local classification and the one or more acoustic features may be determined. In some embodiments, the spectral energy classification is provided to a noise suppression system.

Abstract: A robust noise reduction system may concurrently reduce noise and echo components in an acoustic signal while limiting the level of speech distortion. The system may receive acoustic signals from two or more microphones in a close-talk, hand-held or other configuration. The received acoustic signals are transformed to cochlea domain sub-band signals and echo and noise components may be subtracted from the sub-band signals. Features in the acoustic sub-band signals are identified and used to generate a multiplicative mask. The multiplicative mask is applied to the noise subtracted sub-band signals and the sub-band signals are reconstructed in the time domain.

Abstract: An audio signal is processed to derive primary and ambient components of the signal. The signal is first transformed to generate frequency-domain subband signals. Primary and ambient components are separated by comparing frequency subband content using a complex-valued similarity metric, wherein one of the primary and ambient components is determined to be the residual after the other is identified using the similarity metric.

Abstract: A system and method are disclosed for enhancing audio signals by nonlinear spectral operations. Successive portions of the audio signal are processed using a subband filter bank. A nonlinear modification is applied to the output of the subband filter bank for each successive portion of the audio signal to generate a modified subband filter bank output for each successive portion. The modified subband filter bank output for each successive portion is processed using an appropriate synthesis subband filter bank to construct a modified time-domain audio signal. High modulation frequency portions of the audio signal may be emphasized or de-emphasized, as desired. The modification may be applied within one or more frequency bands.

Abstract: The present technology provides a robust noise suppression system which may concurrently reduce noise and echo components in an acoustic signal while limiting the level of speech distortion. An acoustic signal may be received and transformed to cochlear domain sub-band signals. Features such as pitch may be identified and tracked within the sub-band signals. Initial speech and noise models may be then be estimated at least in part from a probability analysis based on the tracked pitch sources. Speech and noise models may be resolved from the initial speech and noise models and noise reduction may be performed on the sub-band signals and an acoustic signal may be reconstructed from the noise-reduced sub-band signals.

Abstract: Processing an audio signal associated with a sound recording made available to be rendered to an end user is disclosed. The audio signal is received. A high-order perceptual attribute of the audio signal as rendered is changed by modifying the audio signal. The modification may be based on real-time analysis of the audio signal.

Abstract: Mapping a plurality of external control signals to values for one or more internal parameters is disclosed. Each external control signal is mapped to a control signal-specific set of internal parameter values corresponding to the level of the signal. The resulting signal-specific sets of internal parameter values are combined to generate a combined set of internal parameter values. The combined set of internal parameter values includes a single value for each internal parameter for which a value was included in at least one signal-specific set of internal parameter values. For each internal parameter for which a value is included in more than one of the signal-specific sets of internal parameter values, the combined set of internal parameter values includes a combined value based at least in part on the values included in the signal-specific sets of internal parameter values.

Abstract: The present technology provides adaptive noise reduction of an acoustic signal using a sophisticated level of control to balance the tradeoff between speech loss distortion and noise reduction. The energy level of a noise component in a sub-band signal of the acoustic signal is reduced based on an estimated signal-to-noise ratio of the sub-band signal, and further on an estimated threshold level of speech distortion in the sub-band signal. In embodiments, the energy level of the noise component in the sub-band signal may be reduced to no less than a residual noise target level. Such a target level may be defined as a level at which the noise component ceases to be perceptible.

Abstract: An array of microphones utilizes two sets of two microphones for noise suppression. A primary microphone and secondary microphone of the three microphones may be positioned closely spaced to each other to provide acoustic signals used to achieve noise cancellation. A tertiary microphone may be spaced with respect to either the primary microphone or the secondary microphone in a spread-microphone configuration for deriving level cues from audio signals provided by tertiary and the primary or secondary microphone. Signals from two microphones may be used rather than three microphones. The level cues are expressed via an inter-microphone level difference (ILD) which is used to determine one or more cluster tracking control signals. The ILD based cluster tracking signals are used to control the adaptation of null-processing noise cancellation modules.