Abstract: A method and system directed to controlling audio devices with active noise reduction. The system detects a first instability condition in a first headphone based on one or more parameters; alters a parameter of the one or more parameters; and detects a second audio instability condition in the first headphone based on the altered parameter of the one or more parameters. The first and second audio instability conditions are related to audio feedback.

Abstract: Methods, systems, and apparatuses are described for optimizing user content consuming experience by recognizing and classifying different sounds while a user views a program. The system may have or may access information related to the program audio being presented, enabling it to distinguish between conversations occurring in the program audio and conversations between users in the viewing environment. The system may turn the program volume down on one or more sound producing devices if it detects a conversation. The system may turn the program volume up if it detects an interrupting noise. The system may also adjust the program content based on locations of various objects within the listening or viewing environment, and types of users in the environment.

Abstract: A method, computer program product, and computing system for obtaining one or more speech signals from a first device, thus defining one or more first device speech signals. One or more speech signals may be obtained from a second device, thus defining one or more second device speech signals. A noise component model may be selected from a plurality of noise component models based upon, at least in part, the one or more first device speech signals and the one or more second device speech signals. The one or more second device speech signals may be augmented, at run-time, based upon, at least in part, the noise component model.

Abstract: A home appliance including a first microphone that is disposed on a surface of a housing, a second microphone that is disposed on an inside of the housing, and a processor configured to perform signal processing for first voice data that is acquired from the first microphone, and perform voice recognition using the signal-processed first voice data. The processor is further configured to generate noise data using second voice data that is acquired from the second microphone and perform signal processing for the first voice data using the generated noise data.

Abstract: A method for filtering out a background audio signal includes: obtaining a first audio signal collected during playing of the background audio signal, the background audio signal being an audio signal obtained by adding watermark information to an original audio signal; separating the first audio signal to obtain the watermark information and a second audio signal without the watermark information; querying the preset correspondence according to the watermark information to obtain the original audio signal corresponding to the watermark information; and filtering out the original audio signal from the second audio signal to obtain a target audio signal.

Abstract: Examples of array geometry agnostic multi-channel personalized speech enhancement (PSE) extract speaker embeddings, which represent acoustic characteristics of one or more target speakers, from target speaker enrollment data. Spatial features (e.g., inter-channel phase difference) are extracted from input audio captured by a microphone array. The input audio includes a mixture of speech data of the target speaker(s) and one or more interfering speaker(s). The input audio, the extracted speaker embeddings, and the extracted spatial features are provided to a trained geometry-agnostic PSE model. Output data is produced, which comprises estimated clean speech data of the target speaker(s) that has a reduction (or elimination) of speech data of the interfering speaker(s), without the trained PSE model requiring geometry information for the microphone array.

Abstract: A noise reduction method and a noise reduction apparatus are provided. The noise reduction method is applied to a keyboard of an integrated terminal device, and the noise reduction method includes: obtaining a first voice source and a second voice source, where the first voice source is a fidelity voice source, the second voice source is an audio signal that includes the first voice source and a noise signal, and the noise signal comes from noise generated by vibration of the keyboard, and/or the noise signal comes from noise of an environment in which the integrated terminal device is located; determining the noise signal based on the first voice source and the second voice source. Based on the technical solutions in this application, the noise signal generated by the integrated terminal device can be offset, so that noise reduction processing is implemented and user experience is improved.

Abstract: Real-time low-complexity stereo speech enhancement with spatial cue preservation may be performed. A stereo speech enhancement system receives a stereo input signal (e.g., a left and right input signal). The stereo speech enhancement system estimates spatial cues for a target speaker and downmixes the stereo input signal into a monaural signal. A low-complexity model may then process the monaural signal to generate an enhanced monaural signal. The stereo speech enhancement system upmixes the enhanced monaural signal based on the estimated spatial cues for the target speaker, to generate an enhanced stereo output signal.

Abstract: Some implementations involve receiving a content stream that includes audio data, receiving at least one type of level adjustment indication relating to playback of the audio data and controlling a level of the input audio data, based on the at least one type of level adjustment indication, to produce level-adjusted audio data. Some examples involve determining, based at least in part on the type(s) of level adjustment indication, a multiband limiter configuration, applying the multiband limiter to the level-adjusted audio data, to produce multiband limited audio data and providing the multiband limited audio data to one or more audio reproduction transducers of an audio environment.

Abstract: A voice processing device includes plural microphones arranged so as to correspond to a plurality of positions. The voice processing device includes at least one memory that stores instructions and voice signals from the plural microphones, and a processor. The voice signals collected by the plural microphones, respectively, during a prescribed period before a present time, are repeatedly stored in the at least one memory as buffered voice signals. The processor detects whether a prescribed word is uttered by a speaker based on the voice signals collected by the plural microphones, determines a microphone corresponding to the speaker by referring to the buffered voice signals, and suppresses the voice signals collected by the plural microphones other than the microphone corresponding to the speaker.

Abstract: The present disclosure may provide a voice audio data processing system. The voice audio data processing system may obtain voice audio data, which includes one or more voices, each being respectively associated with one of one or more subjects. For one of the one or more voices and the subject associated with the voice, the voice audio processing system may generate a text based on the voice audio data. The text may have one or more sizes, each size corresponding to one of one or more volumes of the voice. The text may have one or more colors, each color corresponding to one of one or more emotion types of the voice.

Abstract: A system comprising a feedback device comprising one or more elements operable to generate an output in response to an input from a processing device, a microphone operable to capture one or more audio inputs, an input detection unit operable to detect one or more inputs to the feedback device from the processing device, and a response generation unit operable to generate a response to the detected input in dependence upon the input and at least a distance between an element associated with the corresponding output and the microphone.

Abstract: An apparatus, method and computer program is described comprising: capturing images using one or more imaging devices of a foldable user device, wherein the user device comprises a visual display, wherein, in a first mode of operation, a display output on the visual display is modified depending on a folding angle of the foldable user device; capturing audio using one or more microphones of the foldable user device; providing a wind mode indication in a second mode of operation; and disabling the modification of the visual display depending on the folding angle in the second mode of operation.

Abstract: A camera includes a first microphone, a second microphone, one or more drains, and a processor. The processor determines a correlation metric between portions of audio signals obtained from the first and second microphones. The one or more drains drain water away from the first microphone, the second microphone, or both. The camera includes a memory to store the portions of the audio signals as portions of an output audio signal.

Abstract: Certain embodiments of the disclosure include a first electronic device, comprising at least one microphone, a communication circuit, and at least one processor, wherein the at least one processor is configured to: establish a first communication connection with an external electronic device using the communication circuit, transmit through the first communication connection, to the external electronic device, using the communication circuit, information for a first voice signal obtained using the at least one microphone, receive second noise information for a second voice signal from the second electronic device, by the communication circuit, wherein the second voice signal corresponding to the first voice signal was obtained by a second electronic device while the information for the first voice signal was transmitted to the external electronic device, and transmit, based on first noise information for the first voice signal and the received second noise information satisfy a specified condition, first cont

Abstract: Described is a system for Neuromorphic Adaptive Core (NeurACore) signal processor for ultra-wide instantaneous bandwidth denoising of a noisy signal. The NeurACore signal processor includes a digital signal pre-processing unit for performing cascaded decomposition of a wideband complex valued In-phase and Quadrature-phase (I/Q) input signal in real time. The wideband complex valued I/Q input signal is decomposed into I and Q sub-channels. The NeurACore signal processor further includes a NeurACore and local learning layers for performing high-dimensional projection of the wideband complex valued I/Q input signal into a high-dimensional state space; a global learning layer for performing a gradient descent online learning algorithm; and a neural combiner for combining outputs of the global learning layer to compute signal predictions corresponding to the wideband complex valued I/Q input signal.

Abstract: An audio processing system includes: a first microphone configured to output a first signal based on a first audio signal; one or more microphones each of which outputs a microphone signal based on an audio signal; one or more adaptive filters configured to respectively receive the microphone signals from the one or more microphones and output passing signals based on the microphone signals; and a processor configured to: determine whether the microphone signal includes uncorrelated noise; control one or more filter coefficients of the one or more adaptive filters; and subtract a subtraction signal based on the passing signals from the first signal. The one or more microphones include a second microphone that outputs a second signal. When determining that the second signal includes the uncorrelated noise, the processor is configured to set a level of the second signal input to the corresponding adaptive filter to zero.

Abstract: The present disclosure provides novel systems and methods of suppressing audio leakage in watch-together sessions. In an embodiment, a device is provided comprising a speaker, a microphone, a media player configured to provide a first media content audio, and a peer to peer (“P2P”) communications engine including a local loopback module and a conferencing module. The local loopback module configured to receive the first media content audio, provide a second media content audio, and amplify, using the speaker, the second media content audio, and the conferencing module configured to capture, using the microphone, captured conference audio, the captured conference audio including the second media content audio, and transmit to a modified captured conference audio; wherein the P2P communications engine is configured to generate the modified captured conference audio by suppressing the second content audio from the captured conference audio.

Abstract: A method of real-time noise reduction including generating spectral data using temporally localized spectral representations of a received audio signal, determining detection of voice by comparing first and second filtered data, and generating noise-reduced audio output by attenuating noise based on the determined detection of voice. The first and second filtered data are formed by attenuating temporal variations of the spectral data based on, respectively, a first timescale and a second timescale. A noise reduction system, comprising processing circuitry configured to execute a method of real-time noise reduction to generate an output that is transmitted via an output port of the noise reduction system. A noise-reduction microphone comprising a housing having a transducer coupled to a processor therein to execute a method of real-time noise reduction, and an output port. A non-transitory computer-readable medium having instructions to cause a processor to perform a method of real-time noise reduction.

Abstract: A spatial audio processing system operable to enable audio signals to be spatially extracted from, or transmitted to, discrete locations within an acoustic space. Embodiments of the present disclosure enable an array of transducers being installed in an acoustic space to combine their signals via inverting physical and environmental models that are measured, learned, tracked, calculated, or estimated. The models may be combined with a whitening filter to establish a cooperative or non-cooperative information-bearing channel between the array and one or more discrete, targeted physical locations in the acoustic space by applying the inverted models with whitening filter to the received or transmitted acoustical signals. The spatial audio processing system may utilize a model of the combination of direct and indirect reflections in the acoustic space to receive or transmit acoustic information, regardless of ambient noise levels, reverberation, and positioning of physical interferers.

Abstract: A drone includes a motor, a noise receiver, a camera, a distance measurer, and a directed sound beam generator. The noise receiver is configured to detect a noise caused by the motor. The camera is configured to capture an image of an area when the drone is in the air. The distance measurer is configured to measure a distance between the drone and a particular point in the captured image. The directed sound beam generator is configured to emit a sound beam that is directed to a particular direction. The drone further includes a processor configured to analyze the detected noise to determine a frequency spectrum of the detected noise. The processor is further configured to analyze the captured image to identify a target, and cause the directed sound beam generator to emit a sound beam to actively cancel at least a portion of the noise directed to the target.

Abstract: The present system ensures that external noise sources do not taint sensor data recorded about a user's sleep parameters, to reduce false positives and false negatives, and provides a more accurate record of the user's sleep sounds, by cycling on and off noise sources. Additionally, the present system, in one embodiment, may be used to track changes in sound patterns from the user and from the environment. This may lead to early warning of health issues, as well as issues with the devices in the bedroom.

Abstract: Techniques for managing call sessions in multi-display systems are described. For instance, the described techniques can be implemented to manage media content in the context of a client device that includes two or more housings attached via a hinge region such that the housings are pivotable relative to one another. The described techniques, for example, enable dynamic configuration of output of call sessions based on changes in user position relative to a client device.

Abstract: A method of multi-talker separation using a 3-tuple coprime microphone array, including generating, by a subarray signal processing module, a respective subarray data set for each microphone subarray of the 3-tuple coprime microphone array based, at least in part, on an input acoustic signal comprising at least one speech signal. The input acoustic signal is captured by the 3-tuple coprime microphone array. The 3-tuple coprime microphone array includes three microphone subarrays. The method includes determining, by the subarray signal processing module, a point by point product of the three subarray data sets; and determining, by the subarray signal processing module, a cube root of the point by point product to yield an acoustic signal output data. The acoustic signal output data has an output amplitude and an output phase corresponding to an input amplitude and an input phase of a selected speech signal of the at least one speech signal.

Abstract: A signal e(z) obtained by subtracting echo-canceling sound from an output of a second microphone is used as an error of an echo cancellation adaptive filter and a noise cancellation adaptive filter, and an output of a first sound source device is added to the signal e(z) and output from a first speaker. The echo cancellation adaptive filter generates echo-canceling sound from an addition signal of the output of the second sound source device and the output of the first microphone such that the signal e(z) is minimized. The noise cancellation adaptive filter generates the noise-canceling sound from the output of the first sound source device such that the signal e(z) is minimized, and outputs the noise-canceling sound from the second speaker.

Abstract: An apparatus for providing active noise control, includes: one or more microphones configured to detect sound entering through an aperture of a building structure; a set of speakers configured to provide sound output for cancelling or reducing at least some of the sound; and a processing unit communicatively coupled to the set of speakers, wherein the processing unit is configured to provide control signals to operate the speakers, wherein the control signals are independent of an error-microphone output.

Abstract: A display device including an external device interface configured to connect the display device to an external speaker; a microphone configured to receive a microphone signal; and a controller configured to: transmit an inaudible test signal to the external speaker to measure a signal delay between the external speaker and the display device, and extract a voice signal by removing an acoustic signal output from the external speaker from the microphone signal based on the measured signal delay.

Abstract: A sound recognition apparatus (100) includes at least two microphones (1) that detect a surrounding sound, an attitude detection unit (4) that detects attitudes of the microphones (1), a self-position estimation unit (52) that estimates positions of the microphones (1) based on the attitudes of the microphones (1) and a signal of the sound detected by the microphone (1), a sound source estimation unit (53) that estimates a direction of the sound source (200) of the sound based on the attitudes of the microphones (1) and the signal of the sound detected by the microphone (1), and a sound environment mapping unit (54) that creates a sound environment map based on the positions of the microphones (1) and the direction of the sound source (200), the sound environment map displaying at least a position of the sound source (200).

Abstract: Disclosed are systems and methods for a frontend capture module of a video conferencing application, which can modify an input signal, received from a microphone device to match predetermined signal characteristics, such as voice signal level and expected noise floor. An Input stage, a suppression module and an output stage amplify the voice signal portion of the input signal and suppress the noise signal of input signal to predetermined ranges. The input stage selectively applies gains defined by a gain table, based on signal level of the input signal. The suppression module selectively applies a suppression gain to the input signal based on presence or absence of voice signal in the input signal. The output stage further amplifies the input signal in portions having a voice signal and applies a gain table to maintain a consistent noise floor.

Abstract: Embodiments that translate a sensor measurement to a frequency characteristic are disclosed. The frequency characteristic can be wirelessly detected by a reader device. The detected frequency characteristic can be used to determine the corresponding sensor measurement. Devices utilizing this approach can be characterized or calibrated to increase accuracy. Systems and methods utilizing the approaches are also described.

Abstract: Embodiments include a microphone assembly comprising an array microphone and a housing configured to support the array microphone and sized and shaped to be mountable in a drop ceiling in place of at least one of a plurality of ceiling tiles included in the drop ceiling. A front face of the housing includes a sound-permeable screen having a size and shape that is substantially similar to the at least one of the plurality of ceiling tiles. Embodiments also include an array microphone system comprising a plurality of microphones arranged, on a substrate, in a number of concentric, nested rings of varying sizes around a central point of the substrate. Each ring comprises a subset of the plurality of microphones positioned at predetermined intervals along a circumference of the ring.

Abstract: Disclosed are systems and methods for performing adaptive equalization operations to reduce variations in the frequency response of audio signals at the eardrum of a wearer of an earphone. The assumption that the frequency response of the error microphone matches the frequency response at the eardrum may not be true due to signal leakage effects and the shape of the ear canal. Techniques are disclosed for the adaptive equalization operations to perform a calibration algorithm to compensate for the effects of the signal leakage and the shape of the ear canal. The earphone may estimate the transfer function from the speaker to the error microphone and may estimate the load impedance from the earphone based on a calibrated relationship between the transfer function and the load impedance. The adaptive equalization may use a calibration algorithm to adjust the frequency response at the error microphone to match that at the eardrum.

Abstract: In some implementations, a user device may receive input that triggers transmission of information via sound. The user device may select an audio clip based on a setting associated with the device, and may modify a digital representation of the selected audio clip using an encoding algorithm and based on data associated with a user of the device. The user device may transmit, to a remote server, an indication of the selected audio clip, an indication of the encoding algorithm, and the data associated with the user. The user device may use a speaker to play audio, based on the modified digital representation, for recording by other devices. Accordingly, the user device may receive, from the remote server and based on the speaker playing the audio, a confirmation that users associated with the other devices have performed an action based on the data associated with the user of the device.

Abstract: An audio signal processing method includes obtaining an audio signal of a sound source, performing first filter processing on the audio signal to generate an imaginary sound source of a virtual space, performing second filter processing on the audio signal to adjust a tone of the imaginary sound source, and outputting an initial reflected sound control signal generated by using the audio signal on which the first filter processing and the second filter processing have been performed, in a first case where the first filter processing is performed before the second filter processing, the first filter processing is performed on the audio signal, and the second filter processing is performed on the audio signal on which the first filter processing has been performed, and in a second case where the second filter processing is performed before the first filter processing, the second filter processing is performed on the audio signal, and the first filter processing is performed on the audio signal on which the seco

Abstract: A cochlear stimulation system for determining a Temporal Fine Structure (TFS) parameter of a spectral component of an incoming acoustic signal is provided. The system comprises a transducer receiving an incoming acoustic signal, and wherein a sampled audio signal of length N samples is provided based on the incoming acoustic signal, a storing unit including a plurality of window frequency differences and/or a plurality of window energy level differences. The system further comprises a signal processor for estimating the TFS parameter of the full frequency range of the sampled audio signal by locating a main spectrum sample of the plurality of spectrum samples having a maximum energy level within a range of frequencies centered around the main spectrum sample, and determining the TFS parameter based on the energy level difference for one or more of the spectrum samples of the group of spectrum samples and the window function.

Abstract: A detection device includes a pressure sensor, which provides a pressure signal indicative of an ambient pressure in an operating environment. An electrostatic-charge-variation sensor provides a charge-variation signal indicative of a variation of electrostatic charge associated with the operating environment, and processing circuitry is coupled to the pressure sensor and to the electrostatic-charge-variation sensor so as to receive the pressure signal and the charge-variation signal, and jointly processes the pressure signal and the charge-variation signal for detecting a variation between a first operating environment and a second operating environment for the detection device. The second operating environment is different from the first operating environment.

Abstract: A key word detection apparatus and a method for low-power voice-activated devices are presented. A first signal processing module operates with a first transducer to receive an incoming signal and generates a first sample. A second signal processing module operates with a second transducer which receives an incoming signal and generates a second sample. In summary, a signal processing system, in particular a key word detection system, has a first low power module that wakes up a second higher power module. The second module uses signals from the first module in order to improve accuracy of key word detection or other signal processing tasks.

Abstract: An audio signal processing apparatus is provided by the present disclosure, and includes: multiple microphones; and every two of the multiple microphones being arranged in close proximity to each other, and the multiple microphones forming a symmetrical structure.

Abstract: An acoustic sensor assembly includes a non-directional acoustic sensor having a first directional pattern, a plurality of directional acoustic sensors surrounding the non-directional acoustic sensor and including a plurality of resonators having different resonance frequencies from each other, each of the plurality of directional acoustic sensors having a second directional pattern, and a processor configured to obtain output signals from the non-directional acoustic sensor and the plurality of directional acoustic sensors. The processor is further configured to calculate an acoustic signal having directivity by selecting any one or any combination of the obtained output signals or selectively combining the obtained output signals, and obtain sound around the acoustic sensor assembly, using the calculated acoustic signal.

Abstract: A speech processing device includes a processor. The processor performs operations including: detecting a single-talk state based on a speech signal collected by each of microphones, the single-talk state in which any one of persons speaks; estimating a mixing rate indicating a ratio of a speech signal of the main speaking person to a speech signal of another person based on a sound pressure ratio of the speech signals collected by the microphones in the single-talk state of the main speaking person and a sound pressure ratio of the speech signals collected by the plurality of microphones in the single-talk state of the another person; and determining whether suppression of a crosstalk component due to speaking of the another person contained in the speech signal of the main speaking person is necessary based on an estimation result of the mixing rate.

Abstract: A robot and operation method is disclosed. The robot according to the present disclosure may include a sensor, a microphone, and a controller. The robot may execute an artificial intelligence (AI) algorithm and/or a machine learning algorithm, and may communicate with other electronic devices in a 5G communication environment. An embodiment may include detecting a movement of the robot to a location; detecting an obstacle within a predetermined range from the robot; estimating an occupation area of the obstacle in space; and identifying a sound signal received from the estimated occupation area of the obstacle from among a plurality of sound signals received by a plurality of microphones of the robot at the location.

Abstract: An earphone includes a feedforward microphone located outside ear and a feedback microphone located inside ear.

Abstract: A method, apparatus and device for displaying a lyric, and a storage medium. The method includes: displaying a control panel and a lyric display panel of a target player application on an interface; canceling the display of the control panel when a trigger operation on the interface satisfies a reference condition, and setting the lyric display panel to be in a non-triggerable state, such that a desktop area of the current interface covered by the control panel and the lyric display panel becomes capable of receiving a trigger operation; displaying an interactive control on the interface, the interactive control being used to perform a lyric display control function; and restoring the display of the control panel when a click operation on the interactive control is detected.

Abstract: Techniques for dynamically adjusting received audio are described. In an example, a computer system receives audio data representing noise and utterance received by a device during a first time interval that has a start and an end. The start corresponds to a beginning of the utterance. The end corresponds to at a selection by the device of an audio beam associated with a direction towards an utterance source. The computer system determines a value associated with an audio adjustment factor. The audio adjustment factor is represented by values that vary during the time interval. The value is one of the values associated with a time point of the first time interval. The computer system generates, based at least in part on the audio data and the value, first data that indicates a measurement of at least one of the noise or the utterance.

Abstract: A waveguide assembly for guiding sound includes a chassis that provides a cavity configured to receive sound propagating in a forwards direction along a primary axis of the waveguide assembly; a fixed waveguide that is fixed with respect to the chassis and positioned on the primary axis of the waveguide assembly, wherein the fixed waveguide is spaced apart from the chassis and is configured to guide sound received by the cavity through at least one opening formed between the fixed waveguide element and the chassis; a moveable waveguide that is moveable with respect to the chassis between: a standby position in which the moveable waveguide is configured to obstruct the at least one opening and form a forward-facing surface of the waveguide assembly; an operational position in which the moveable waveguide is configured to allow sound to exit the cavity through the at least one opening.

Abstract: The present disclosure describes aspects of adaptive energy limiting for transient noise suppression. In some aspects, an adaptive energy limiter sets a limiter ceiling for an audio signal to full scale and receives a portion of the audio signal. For the portion of the audio signal, the adaptive energy limiter determines a maximum amplitude and evaluates the portion with a neural network to provide a voice likelihood estimate. Based on the maximum amplitude and the voice likelihood estimate, the adaptive energy limiter determines that the portion of the audio signal includes noise. In response to determining that the portion of the audio signal includes noise, the adaptive energy limiter decreases the limiter ceiling and provides the limiter ceiling to a limiter module effective to limit an amount of energy of the audio signal. This may be effective to prevent audio signals from carrying full energy transient noise into conference audio.

Abstract: An open acoustic device (100) may include a fixing structure (120) configured to fix the acoustic device (100) near an ear of a user without blocking an ear canal of the user; a first microphone array (130) configured to acquire environmental noise (410); a signal processor (140) configured to: determine, based on the environmental noise, a primary route transfer function (420) between the first microphone array (130) and the ear canal of the user; estimate, based on the environmental noise and the primary route transfer function, a noise signal (430) at the ear canal of the user; and generate, based on the noise signal at the ear canal of the user, a noise reduction signal (440); and a speaker (150) configured to output, according to the noise reduction signal, a noise reduction acoustic wave (450), the noise reduction acoustic wave being configured to eliminate the noise signal.

Abstract: Apparatus, methods and computer-readable medium are provided for processing wind noise. Audio input is processed by receiving an audio input. A wind noise level representative of a wind noise at the microphone array is measured using the audio input and a determination is made, based on the wind noise level, whether to perform either (i) a wind noise suppression process on the audio input on-device, or (ii) the wind noise suppression process on the audio input on-device and an audio reconstruction process in-cloud.

Abstract: Techniques for improving microphone noise detection and suppression are provided. A method for detecting wind noise corresponding to microphone signals may include determining a phase of a complex coherence of the microphone signals. The phase may be used to determine a presence of wind near the microphones. A derivative of the phase may be used to determine a presence of speech near the microphones. Further, a method for suppressing noise caused by the wind may include determining a gain based on a cross power spectrum of the microphone signals and applying the gain to the microphone signals. The method for suppressing the noise may also include attenuating the microphone signals using a post filter. Based on detection of the wind, microphones which are more exposed to the wind may not be used to process the speech whereas microphones less exposed to the wind may be used to process the speech.

Abstract: An apparatus and method for auto echo cancellation utilizing an occluded voice sensor. The technology as disclosed and claimed herein and the various implementations and embodiments improves Acoustic Echo Cancellation (AEC) system performance by increasing cancellation quality and speech SNR by the replacement of the lower frequency portion of the microphone signal with the signal of an Occluded Voice Sensor (OVS) signal and never including the spectral band in the transmission. This frequency band replacement excludes the reinjection of that band from the speaker.