Abstract: Disclosed are methods and devices for communicating between a user device, such as a mobile terminal, and a three wire headset assembly having two microphones coupled thereto. The methods provide two microphone uplink communication and stereo audio playback on a three wire headset assembly without requiring a fourth wire or the addition of a separate voice processor. The headset assembly includes a primary microphone and secondary microphone spaced some distance apart for applying techniques of noise reduction and suppression. The headset assembly includes two speakers. Controlled switches are also included in both the user device and headset assembly to achieve dual use for a single wire of the three wires. The three wire headset may advantageously support both stereo playback and two microphone voice processing. In some embodiments, two microphone noise suppression for a wired headset mode of the user device is provided.

Abstract: Systems and methods for envelope-based acoustic echo cancellation in a communication device are provided. In exemplary embodiments, a primary acoustic signal is received via a microphone of the communication device, and a far-end signal is received via a receiver. Frequency analysis is performed on the primary acoustic signal and the far-end acoustic signal to obtain frequency sub-bands. An echo gain mask based on magnitude envelopes of the primary and far-end acoustic signals for each frequency sub-band is generated. A noise gain mask based on at least the primary acoustic signal for each frequency sub-band may also be generated. A combination of the echo gain mask and noise gain mask may then be applied to the primary acoustic signal to generate a masked signal. The masked signal is then output.

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 modification of an audio input signal are provided. In exemplary embodiments, an adaptive multiple-model optimizer is configured to generate at least one source model parameter for facilitating modification of an analyzed signal. The adaptive multiple-model optimizer comprises a segment grouping engine and a source grouping engine. The segment grouping engine is configured to group simultaneous feature segments to generate at least one segment model. The at least one segment model is used by the source grouping engine to generate at least one source model, which comprises the at least one source model parameter. Control signals for modification of the analyzed signal may then be generated based on the at least one source model parameter.

Abstract: Provided are methods and systems for enhancing the quality of voice communications. The method and corresponding system may involve classifying an audio signal into speech, and speech and noise and creating speech-noise classification data. The method may further involve sharing the speech-noise classification data with a speech encoder via a shared memory or by a Least Significant Bit (LSB) of a Pulse Code Modulation (PCM) stream. The method and corresponding system may also involve sharing acoustic cues with the speech encoder to improve the speech noise classification and, in certain embodiments, sharing scaling transition factors with the speech encoder to enable the speech encoder to gradually change data rate in the transitions between the encoding modes.

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: 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: An audio device performs self calibration with respect to an audio source location when processing an audio signal frame determined likely to be dominated by the audio source. One or more conditions for sub-bands within the audio frame are evaluated to help identify whether the frame is dominated by the audio source. If the conditions meet a threshold value for a number of sub-bands within the frame, the audio signal may be identified as one dominated by the desired audio source and an audio source location coefficient may be adapted. Additionally, when the audio source location coefficient falls below a threshold value, (e.g., suggesting that one of two or more microphones is blocked), noise suppression is reduced or eliminated for the frame or frame sub-bands to prevent suppression of a desired audio source component along with the noise component.

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 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 blind subband acoustic echo cancellation postfiltering in a communication device are provided. In exemplary embodiments, an acoustic signal is received via a microphone of the communication device. Acoustic echo cancellation (AEC) is applied to the acoustic signal to obtain an AEC masked signal. Because residual echo may still exist in the AEC masked signal, blind subband AEC postfiltering on the AEC masked signal may be performed to obtain an echo-free acoustic signal. The echo-free signal may then 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 audio signal processing are provided. In exemplary embodiments, a filter cascade of complex-valued filters are used to decompose an input audio signal into a plurality of frequency components or sub-band signals. These sub-band signals may be processed for phase alignment, amplitude compensation, and time delay prior to summation of real portions of the sub-band signals to generate a reconstructed audio signal.

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: The present technology measures distortion introduced by a noise suppression system. The distortion may be measured as the difference between a noise-reduced speech signal and an estimated idealized noise reduced reference (EINRR). The EINRR may be determined from a speech component and noise component that are pre-processed, and the EINRR may be used with masks associated with energies lost and added in the speech component and noise component. The EINRR may be calculated on a time varying basis.

Abstract: The present technology measures distortion introduced by a noise suppression system. The distortion may be measured as the difference between a noise-reduced speech signal and an estimated idealized noise reduced reference (EINRR). The EINRR may be determined from a speech component and noise component that are pre-processed, and the EINRR may be used with masks associated with energies lost and added in the speech component and noise component. The EINRR may be calculated on a time varying basis.

Abstract: A method of removing reverberation from audio signals is disclosed. The method comprises spectro-temporally analyzing the first audio signal and the second audio signal to derive an energy function of time for a plurality of frequency bands. The method further comprises determining a delay stability between the energy function of time for the first audio signal and the second audio signal in each band, determining a gain function in each band based on the delay stability, adjusting the energy of the first audio signal and the second audio signal using the gain function within each band, and resynthesizing audio signals from the energy in each band of the first audio signal and the second audio signal.

Abstract: A system and method are disclosed for processing an audio signal including separating the audio signal into a plurality of streams which group sounds from a same source prior to classification and analyzing each separate stream to determine phoneme-level classification. One or more words of the audio signal may then be outputted.

Abstract: A system for processing audio signals comprises a sequence of digital filters each configured to process a selected frequency using a set of coefficients. A filter configured to process a certain frequency shares its coefficients with another filter that processes a frequency that is lower than the first frequency by at least one frequency interval, such as an octave. The first filter samples at a certain sampling rate, and the second filter's sampling rate is determined by multiplying the first sampling rate by the ratio of the second frequency to the first frequency. The filters are evenly grouped into frequency intervals, such as octaves. Filters in an octave are sampled at a sampling frequency that is at least twice as high as the highest frequency processed in that octave.