Abstract: A method for eliminating acoustic reverberation in an audio signal. First and second level measurements are performed on the audio signal. The first level measurement uses a first attack parameter and a first decay parameter to measure a first attack time and a first decay time. The second level measurement uses a second attack parameter and a second decay parameter to form a second attack time that is identical to the first attack time and a second decay time that is longer than the first decay time. A difference between the first and second level measurements is calculated. The difference and the second level measurement are used to estimate a reverberation interference level, and the first level measurement and the reverberation interference level are used to ascertain a gain parameter for the audio signal.

Abstract: Environmental conditions affecting a sensor having a thermal coefficient are compensated by applying an adaptive filter to an environmental condition reference signal. The resulting adaptive cancellation signal may be used to provide feedback control to a first heating element.

Abstract: A voice call is established between two or more computing devices. The voice call may be, for example, audio only or audio and video. One computing device, referred to as a near-end device, receives a digital representation of background noise captured at another computing device, referred to as a far-end device. The near-end device captures voice audio from the user of the near-end device and adjusts the voice audio based at least in part on the digital representation of the background noise at the far-end device. The near-end device then transmits the adjusted voice audio to the far-end device for playback at the far-end device.

Abstract: A system integrated in a smart TV that suppresses in real-time a selected voice of a specific person in a digital audio stream that includes an audio processor, a setting module, a user interface, and an input mechanism. The system selects the voice of the specific person, creates a voice profile of that voice that includes voice parameters, stores the voice profile, continuously monitors the voice parameters in digital audio stream received by the audio processor, suppresses identified voices that match those parameters o in the received digital audio stream, and send a clean digital audio stream to be played through the speakers of the smart TV. The input mechanism may select the voice by using a key when the voice is heard through the speakers, may assign identification to the selected voice, and may remove the selected voice stored in the audio processor.

Abstract: An oversampling filter oversamples a digital audio signal. A ?? modulator delta-sigma modulates a signal output from the oversampling filter. A D/A converter converts a signal output from the ?? modulator into an analog audio signal. The oversampling filter includes a processor configured to run firmware and a computational algorithm is configurable based on the firmware.

Abstract: The present disclosure is directed to improvements in interference mitigation for Adjacent Channel Leakage in wireline communication, and more specifically, but not exclusively, to improved kernel designs that can facilitate interference mitigation for Adjacent Channel Leakage in cable modem systems. Examples of the present disclosure provide an apparatus for a transceiver device that comprises interference cancellation circuitry configured to cancel interference caused by upstream signals in one or more upstream sub-bands on one or more downstream sub-bands based on a combination of a plurality of kernels. The interference is at least partially caused by non-linearities within a transmission circuitry of the transceiver device, the plurality of kernels representing the non-linearities within the transmission circuitry of the transceiver device. Each of the kernels comprises one or more associated terms, with each of the associated terms being in-band for at least one of the one or more downstream sub-bands.

Abstract: A calibrating device can mitigate the static mismatch error of a digital-to-analog converter (DAC), and includes a digital code generating circuit, the DAC, an analog-to-digital converter (ADC), a filter circuit, an indicating circuit, and a statistical circuit. The digital code generating circuit generates a digital code of N digital codes. The DAC generates an analog signal corresponding to one of N signal levels according to the digital code. The ADC generates a digital signal according to the analog signal. The filter circuit generates a gradient value according to the difference between the digital code and the digital signal. The indicating circuit generates a selection signal according to the digital code. The statistical circuit learns from the selection signal that the gradient value is corresponding to a Kth digital code of the N digital codes, and determines whether the Kth digital code should be adjusted according to the gradient value.

Abstract: A wireless communication device can include a transmitter subsystem configured to transmit a transmit signal that, once propagated from the wireless communication device, may be reflected back and received by a receiver subsystem as interference. The wireless communication device can include a self-interference cancellation subsystem configured to generate a cancellation signal to mix with received signals to mitigate self-interference effects. A performance floor for the self-interference cancellation subsystem may be determined based on a phase noise profile of an oscillator of either or both the transmitter subsystem or the receiver subsystem. The performance floor metric can be thereafter used to inform an operation or operational setting of the wireless communication device.

Abstract: A method for Short-Time Fourier Transform-based echo muting includes receiving a microphone signal including acoustic echo captured by a microphone and corresponding to audio content from an acoustic speaker, and receiving a reference signal including a sequence of frames representing the audio content. For each frame in a sequence of frames, the method includes processing, using an acoustic echo canceler configured to receive a respective frame as input to generate a respective output signal frame that cancels the acoustic echo from the respective frame, and determining, using a Double-talk Detector (DTD), based on the respective frame and the respective output signal frame, whether the respective frame includes a double-talk frame or an echo-only frame. For each respective frame that includes the echo-only frame, muting the respective output signal frame, and performing speech processing on the respective output signal frame for each respective frame that includes the double-talk frame.

Abstract: A sound source tracking method adapted to an ongoing video conference comprising: obtaining a streaming signal of the video conference from an internet; performing a video conference procedure to obtain an audio signal from the streaming signal and send the audio signal to a speaker; performing an audio tracking procedure to obtain the audio signal outputted from the video conference procedure to the communication device and send the audio signal to a sound source tracking camera; playing the audio signal to generate a far-end sound; recording a field sound comprising at least one of the far-end sound and a local-end sound; and performing a comparing procedure to determine a shooting direction of the sound source tracking camera, wherein the shooting direction is adjusted so as not to shoot the speaker when a similarity of the far-end sound and the audio signal is greater than a threshold.

Abstract: A transmitter circuit receives a PWM input signal and a clock signal. A logic circuit generates a control signal as a function of the clock signal. The control signal is normally set to high, and is periodically set to low for a transmission time interval when an edge is detected in the clock signal. The transmission time interval is shorter than a half clock period of the clock signal. A tri-state transmitter receives the PWM input signal and the control signal, and produces first and a second output signals at first and second transmitter output nodes, respectively. The output signals have a voltage swing between a positive voltage and a reference voltage. An output control circuit is sensitive to the control signal and is coupled to the first and second transmitter output nodes.

Abstract: Techniques for improving the power consumption of a device without impacting or with minimal impact to operations of the device are described. In an example, the device includes a processor. The device receives, while the processor is operating in a first power mode, first input data corresponding to first audio detected by a microphone. Based at least in part on the first input data, the device detects a sound event or ambient noise. Based at least in part on a detection of the ambient noise only, the device causes processor to operate in a second power mode in which the processor consumes less power than in the first power mode.

Abstract: A system for detecting three-way calls in a monitored telephone conversation includes a speech recognition processor that transcribes the monitored telephone conversation and associates characteristics of the monitored telephone conversation with a transcript thereof, a database to store the transcript and the characteristics associated therewith, and a three-way Call detection processor to analyze the characteristics of the conversation and to detect therefrom the addition of one or more parties to the conversation. The system preferably includes at least one domain-specific language model that the speech recognition processor utilizes to transcribe the conversation. The system may operate in real-time or on previously recorded conversations. A query and retrieval system may be used to retrieve and review call records from the database.

Abstract: Facilitating echo cancellation within communication networks is contemplated, such as but not necessarily limited to facilitating echo cancellation within full-duplex (FDX) communication networks. The echo cancellation may optionally be performed with an echo canceller included as part of or otherwise associated with an FDX node used to facilitate interfacing signaling between a digital domain and an analog domain of a FDX or other communication network.

Abstract: Disclosed is a voice activity detection (VAD) device and method capable of referring to an environment detection result and thereby selecting one of multiple VAD results as a basis for determining whether a voice activity occurs. The VAD device includes an environment detection circuit, a VAD circuit, and a voice activity decision circuit. The environment detection circuit is configured to process an audio input signal and thereby generate an environment detection result. The VAD circuit is configured to analyze the audio input signal with multiple VAD algorithms and thereby generate multiple VAD results. The voice activity decision circuit is configured to select one of the multiple VAD results according to the environment detection result.

Abstract: In one aspect, a playback deice is configured to identify in an audio stream, via a second wake-word engine, a false wake word for a first wake-word engine that is configured to receive as input sound data based on sound detected by a microphone. The first and second wake-word engines are configured according to different sensitivity levels for false positives of a particular wake word. Based on identifying the false wake word, the playback device is configured to (i) deactivate the first wake-word engine and (ii) cause at least one network microphone device to deactivate a wake-word engine for a particular amount of time. While the first wake-word engine is deactivated, the playback device is configured to cause at least one speaker to output audio based on the audio stream. After a predetermined amount of time has elapsed, the playback device is configured to reactivate the first wake-word engine.

Abstract: A method of detecting human voice activity includes determining a presence of human voice in a frame of audio signal using a plurality of features extracted from the frame of audio signal. The extracted features can include a number of zero-crossings, a periodicity metric, an energy ratio between a low frequency band and a high frequency band, and an envelope-to-floor ratio (EFR) in the frame of audio signal. Each of the features is associated with predefined criteria indicative of a presence of human voice, and based on comparisons of the features to the respective predefined criteria, the voice activity detector determines whether the frame of audio signal includes a human voice.

Abstract: A method includes: generating a first signal according to a digital signal; filtering the first signal according to first filter coefficients of first filter to generate a second signal; adding a first reference signal with the second signal to generate a third signal; performing digital-to-analog conversion according to the first and third signals to generate and output an echo signal; performing analog-to-digital conversion according to the echo signal to generate a fourth signal; generating a fifth signal according to the digital signal and the fourth signal; and updating the first filter coefficients according to the fifth signal.

Abstract: Examples relate to a transmit apparatus, a receive apparatus, a method for transmitting, a method for receiving and computer readable storage media for transmitting and/or receiving in a communication network. A transmit apparatus for transmitting discontinuous time-frequency operation, DTFO, signals in a communication network comprises a transmitter configured to transmit a DTFO signal comprising monitoring symbols and regular DTFO symbols in the communication network.

Abstract: A speaker system includes a wearable speaker capable of outputting a first sound which is a voice of a communication partner of a talker and a second sound, a microphone, and a sound processing device which processes a sound output from the wearable speaker and a sound picked up by the microphone. The sound processing device generates a reference signal by synthesizing a first signal indicating the first sound and a second signal indicating the second sound, outputs the first signal and the second signal to the wearable speaker, obtains a sound pickup signal including the voice of the talker from the microphone, performs, on the sound pickup signal, a process of cancelling the sound component output from the wearable speaker by using the reference signal, and outputs the sound pickup signal on which the cancellation process has been performed.

Abstract: Some disclosed teleconferencing methods may involve detecting a howl state during a teleconference. The teleconference may involve two or more teleconference client locations and a teleconference server. The teleconference server may be configured for providing full-duplex audio connectivity between the teleconference client locations. The howl state may be a state of acoustic feedback involving two or more teleconference devices in a teleconference client location. Detecting the howl state may involve an analysis of both spectral and temporal characteristics of teleconference audio data. Some disclosed teleconferencing methods may involve determining which client location is causing the howl state. Some such methods may involve mitigating the howl state and/or sending a howl state detection message.

Abstract: An echo canceller system includes a data transmitter circuit and an echo canceller circuit. The data transmitter circuit is configured to receive a transmitted signal. The echo canceller circuit includes a first filter. The first filter is configured to generate a first filtered signal according to the transmitted signal and a filter coefficient vector. The filter coefficient vector is updated according to a high-frequency leakage process. The echo canceller circuit is further configured to generate an echo cancelling signal according to the first filtered signal. The data transmitter circuit is further configured to generate an output signal according to a received signal and the echo cancelling signal.

Abstract: Approaches provide for processing a communications signal, such as a wideband communications signal. For example, a communications signal can be received at an access device, such as a cable modem or a wireless transceiver/radio. The access device can include components operable to process the communications signal to enable access to, for example, video, data, voice, high-speed Internet services. At least some of the components include an echo cancellation component among other components to cancel interference (e.g., ACI, ALI). For example, the echo cancellation component can use trained filters and samples of the upstream signal, echoed samples of the transmitted signal, feedback signals of the echoed signal, etc., to estimate one or more types of interference. Thereafter, the estimated interference can be used to cancel interference at the access device.

Abstract: A system for detecting parasitic oscillation in a wearable audio device that includes an electro-acoustic transducer that is configured to develop sound for a user, a housing that holds the transducer, at least one of a feedforward microphone that is configured to detect sound outside of the housing and output a feedforward microphone signal and a feedback microphone that is configured to detect sound inside of the housing and output a feedback microphone signal, and an opening in the housing that emits sound pressure from the transducer. The system includes a parasitic oscillation detector that is configured to determine a fundamental frequency of at least one of the feedforward and feedback microphone signals and compare an amplitude of the determined fundamental frequency to a threshold level, to determine parasitic oscillation.

Abstract: A method includes receiving audio data corresponding to an utterance spoken by a user that includes a command for a digital assistant to perform a long-standing operation, activating a set of one or more warm words associated with a respective action for controlling the long-standing operation, and associating the activated set of one or more warm words with only the user. While the digital assistant is performing the long-standing operation, the method includes receiving additional audio data corresponding to an additional utterance, identifying one of the warm words from the activated set of warm words, and performing speaker verification on the additional audio data. The method further includes performing the respective action associated with the identified one of the warm words for controlling the long-standing operation when the additional utterance was spoken by the same user that is associated with the activated set of one or more warm words.

Abstract: The present application relates to a system and method for providing autoregressive based residual echo suppression in the STFT domain in a bidirectional vehicle communications system including receiving, from a communications processor, a speaker signal for coupling to a speaker, receiving, from a microphone, a microphone signal wherein the microphone signal includes a voice signal and a residual echo signal, transforming the signals to the STFT domain generating an estimated power spectral density of the residual echo signal in response to a prior power spectral density of a prior residual echo signal, isolating the voice signal from the microphone signal by multiplying the estimated residual echo gain generated using the estimated power spectral density of the residual echo signal with the microphone signal, transforming the signals back to the time domain and coupling the voice signal to the communications processor.

Abstract: An audio codec circuit includes a voltage detecting circuit, an output processing circuit, a digital-to-analog conversion circuit, and an audio amplifying circuit. The voltage detecting circuit is configured to detect an input voltage of an input power. The output processing circuit obtains a first output compensation value according to the input voltage, an output circuit parameter, and a first output spec. The output processing circuit processes a digital audio and compensates the processed audio by the first output compensation value. The digital-to-analog conversion circuit is configured to perform digital-to-analog conversion on the compensated audio to obtain an analog audio. The audio amplifying circuit is configured to amplify the analog audio and output the amplified audio.

Abstract: A device implementing a system for processing speech in an audio signal includes at least one processor configured to receive an audio signal corresponding to at least one microphone of a device, and to determine, using a first model, a first probability that a speech source is present in the audio signal. The at least one processor is further configured to determine, using a second model, a second probability that an estimated location of a source of the audio signal corresponds to an expected position of a user of the device, and to determine a likelihood that the audio signal corresponds to the user of the device based on the first and second probabilities.

Abstract: Methods and systems for signal processing an audio signal in a hearing device to detect when feedback path change occurs and controlling an adaptive feedback canceler to remove the feedback are provided. The hearing device includes a receiver and a microphone. An exemplary method includes detecting whether a tonal signal is caused by a feedback path change by estimating a product of a subband error signal and a subband output signal generated in response to a subband audio input signal; estimating a fast metric based on the estimated product and estimating a slow metric; and applying or maintaining an adaptation rate to the adaptive feedback canceler of the hearing device, wherein the adaptation rate applied or maintained is selected based upon a value of the difference between the fast and slow metrics compared to a threshold value.

Abstract: Provided are an echo delay time estimation method and system thereof, wherein the echo delay time estimation method is executed by the echo delay time estimation system with the following steps: receiving a testing signal and a received signal and executing a time to frequency analysis to generate a testing signal spectrogram and a received signal spectrogram; respectively executing a characteristic signal dynamic detection calculation for the testing signal spectrogram and the received signal spectrogram to generate a testing signal characteristic dynamic vector and a received signal characteristic dynamic vector; executing a cross-correlated vector estimation for the testing signal characteristic dynamic vector and the received signal characteristic dynamic vector to generate a cross-correlated vector; and calculating an echo delay time according to the cross-correlated vector.

Abstract: A signal processing device includes an echo estimation device, a captured signal buffer device and a delay estimation device. The echo estimation device generates an echo estimation signal according to a reference signal and a set of reflection path simulation coefficients and compensates the echo estimation signal according to a first delay to generate a compensated echo estimation signal. The captured signal buffer device buffers a captured signal captured by microphone device and outputs the captured signal according to a second delay to generate a compensated captured signal. The delay estimation device estimates an amount of delay adjustment according to the compensated echo estimation signal and the compensated captured signal and updates the first delay or the second delay according to the amount of delay adjustment. A difference between an upper bound and a lower bound of the first delay is smaller than or equal to 1.

Abstract: A method for operating a beamforming microphone array for use in a predetermined area is provided herein, the method comprising: receiving acoustic audio signals at each of a plurality of microphones, converting the same to an electrical mic audio signal, and outputting each of the plurality of electrical mic audio signals; generating a user location data signal by a wave sensor system, and outputting the user location data signal, wherein the user location data signal includes location information of one or more people within the predetermined area; receiving both the user location data signal and plurality of echo-corrected mic audio signals at an adaptive beamforming device; and adapting one or more beams by the adaptive beamforming device based on the user location data signal and plurality of mic audio signals wherein each of the one or more beams acquires sound from one or more specific locations in the predetermined area.

Abstract: A system configured to improve double-talk detection. The system inputs microphone signals into an adaptive filter and determines whether double talk is present based on how the adaptive filter adapts to the microphone signals. For example, when an audible sound is detected, the adaptive filter updates the filter coefficients that correspond to a time difference of arrival of the audible sound. Thus, the device may detect single-talk conditions (e.g., a single peak in the filter coefficients) or double-talk conditions (e.g., two peaks in the filter coefficients). In addition, the device may track a location of the local speech or remote speech over time based on the time difference of arrival.

Abstract: An acoustic processing method includes extracting a feature amount from a first audio signal representing sound collected by a microphone, receiving a second audio signal from a different device on a far-end side, determining a gain of the second audio signal based on the extracted feature amount, and adjusting the second audio signal based on the determined gain of the second audio signal.

Abstract: A transceiver circuit includes an ADC and an echo-cancellation circuit, wherein the echo-cancellation circuit includes a steady circuit, a transient circuit and an output circuit. In the operations of the transceiver circuit, the ADC is configured to perform an analog-to-digital conversion operation on an analog input signal to generate a digital input signal. The steady circuit is configured to generate a steady echo response according to a transmitting signal. The transient circuit is configured to generate an echo response adjustment signal according to a phase change of a clock signal used by the transmitting signal. The output circuit is configured to generate an output signal according to the digital input signal, the steady echo response, and the echo response adjustment signal.

Abstract: A system and method that enhances spoken utterances and provides entertainment by capturing one or more microphone signals containing echo and decomposing the one or more microphone signals into a plurality of signal paths through a synthesizer that adds or makes non-linear modifications to some of the captured one or more microphone signals. The system and method and estimates multiple echo paths from each of the one the one or more microphones. The system and method processes the captured microphone signals in response to the estimated plurality of echo paths by subtracting the echo contributions of each of the plurality of echo paths from the captured one or more microphone signals. The system and method also provide signal separation and post processing functions that renders speech recognition gaming applications.

Abstract: Audio systems and methods are provided that enhance a portion of audio content relative to other portions of the audio content. The systems and methods select the portion to be enhanced and calculate an intelligibility metric of the selected portion, such as a dialogue portion. The systems and methods determine a gain based at least in part upon the intelligibility metric and apply the gain to the selected portion to provide an enhanced portion. The systems and methods provide an audio signal, based at least in part upon the enhanced portion, to an output for conversion to an acoustic signal, such as by an acoustic transducer.

Abstract: A bias compensation method, system, and computer program product include modifying a behavior of a first analytic engine service with a second analytic engine service, where the first service accepts user submitted data and communicates an assessment of the data in a form of a label associated with the corresponding submitted data, where the second service accepts an input and communicates an assessment in a form of a label associated with the corresponding input, and where a behavior model of the first service and the second service includes a discrepancy between the output labels by each service with respect to true labels of data accepted, further including composing a new analytic engine service from the first service and the second service to optimize a service bias in terms of a test dataset based on the behavior model and the known true assessments.

Abstract: Wireless communications systems and methods related to mitigating maximum permissible exposure (MPE) constraints based on user equipment (UE) feedback are provided. A first wireless communication device transmits, to a second wireless communication device, a plurality of reports, each report indicating an allowable transmission power level at the first wireless communication device satisfying an MPE parameter. The first wireless communication device receives, from the second wireless communication device, a first configuration based on an MPE profile associated with the first wireless communication device in response to the plurality of reports. The first wireless communication device transmits, to the second wireless communication device, a first communication signal over a first beam based on the first configuration.

Abstract: Voice control in a multi-talker and multimedia environment is disclosed. In one aspect, there is provided a method comprising: receiving a microphone signal for each zone in a plurality of zones of an acoustic environment; generating a processed microphone signal for each zone in the plurality of zones of the acoustic environment, the generating including removing echo caused by audio transducers in the acoustic environment from each of the microphone signals, and removing interference from each of the microphone signals; and performing speech recognition on the processed microphone signals.

Abstract: The method includes transmitting by a first remote communication unit an upstream symbol with a first structure onto a first communication line at a reference time point trf, wherein the reference time point trf is determined based on a time of reception of a downstream symbol with the first structure tFDX_DS_RX and a first propagation delay over the first communication line tPD1, as trf=tFDX_DS_RX?tPD1; transmitting by a second remote communication unit an upstream symbol with a second structure onto the second communication line at tTDD_US_TX=trf?tPD2 during a time interval assigned for upstream transmission on the second communication line, wherein tPD2 is a second propagation delay over the second communication line, so that the upstream symbol with the second structure transmitted by the second remote communication unit arrives at the access node at the reference time point trf.

Abstract: Automatic equalization for consistent headphone may take place in a playback mode of operation in which ANC is turned off and there is no direct feedback from an internal microphone (to the input of a speaker). An automatic user content equalization process is active during that mode of operation which adapts a filter AEQ to restore a flat or other desired frequency response at the output of the speaker despite variation in headphone fit. An estimate of a transfer function of a path S is determined, wherein the path S is from i) the input of the speaker of the headphone to the internal microphone signal. The filter AEQ is adapted based on the estimate of the transfer function of the path S while it filters user content audio that drives the input of the speaker of the headphone. Other embodiments are also described and claimed.

Abstract: A method for improving voice quality is provided herein. The method includes receiving acoustic signals from a microphone array; receiving sensor signals from an accelerometer sensor of the headset; generating, by a beamformer, a speech output signal and a noise output signal according to the acoustic signals; best-estimating the speech output signal according to the sensor signals to generate a best-estimated signal; and generating a mixed signal according to the speech output signal and the best-estimated signal.

Abstract: Systems and methods for detecting speech activity. The system includes an audio source and an electronic processor. The electronic processor is configured to receive a first audio signal from the audio source, buffer the first audio signal, add random noise to the buffered first audio signal, and filter the first audio stream to create a filtered signal. The electronic processor then determines a signal entropy of each frame of the filtered signal, determines an average signal entropy of a first plurality of frames of the filtered signal occurring at a beginning of the filtered signal, and compares the signal entropy of each frame of the filtered signal to the average signal entropy. Based on the comparison, the electronic processor determines a first speech endpoint located in a first frame of the filtered signal.

Abstract: A system that performs automatic gain control (AGC) using different decay rates. The system may select a slow decay rate to track a loudness level within speech (e.g., within an utterance), improving audio quality and maintaining dynamic range for an individual voice, while selecting a fast decay rate to track the loudness level after a gap of silence (e.g., no voice activity detected for a duration of time) or during large level changes (e.g., actual speech loudness is lower than estimated speech loudness for a duration of time). This improves an accuracy of the loudness estimate and therefore a responsiveness of the automatic gain control, resulting in an improved user experience.

Abstract: Apparatuses, methods, systems, and program products are disclosed for detecting and correcting audio echo. An apparatus includes a processor and a memory that stores code executable by the processor. The code is executable by the processor to detect audio that is received through an audio channel of a first user during a conference call being echoed through an audio channel of a second user in the conference call. The code is executable by the processor to identify the second user associated with the audio channel that is echoing the audio of the first user. The code is executable by the processor to perform a corrective action associated with the second user to prevent audio that is received from the first user from being echoed through the audio channel of the second user.

Abstract: A method for noise cancellation, the method may include detecting a voice trigger or receiving an indication regarding an occurrence of a voice trigger; searching, based at least on a timing of occurrence of the voice trigger, for at least one of a noise-period and a voice-period; when finding a noise period then updating one or more noise parameters based on one or more features of sensed audio signals received during the noise period; when finding a voice period then updating one or more voice parameters based on one or more features of sensed audio signals received during the voice period; estimating, based on the one or more noise parameters and the one or more voice parameters, a probability that voice is present at one or more subsequent periods; and cancelling noise at the one or more subsequent periods, based on the voice presence probability.

Abstract: A beamforming microphone array is described herein, comprising: a plurality of microphones each of which is adapted to receive an acoustic audio signal and convert the same to a microphone (mic) audio signal; a wave sensor system adapted to determine locations of one or more people within a predetermined area about the beamforming microphone array and output the same as user location data signal; and an adaptive beamforming circuit adapted to receive the user location data signal and plurality of mic audio signals and perform adaptive beamforming on the plurality of mic audio signals that takes into account the received user location data signal to adapt a plurality of beam signals, one for each of the microphones, to acquire sound from one or more specific locations in the predetermined area; and a plurality of acoustic echo cancellation devices, one for each of the beam signal outputs from the adaptive beamforming circuit, wherein each of the plurality of acoustic echo cancellation devices is adapted to rec

Abstract: An audio system, including an accelerometer positioned to produce an accelerometer signal representative of road noise within a vehicle cabin; a microphone disposed within the vehicle cabin such that the microphone receives the road noise and produces a microphone signal having a road-noise component; and a road-noise canceler, comprising a road-noise cancellation filter, configured to receive the accelerometer signal and the microphone signal and to minimize the road-noise component of the microphone signal according to the accelerometer signal, to produce an estimated microphone signal.

Abstract: Facilitating echo cancellation within communication networks is contemplated, such as but not necessarily limited to facilitating echo cancellation within full-duplex (FDX) communication networks. The echo cancellation may optionally be performed with an echo canceller included as part of or otherwise associated with an FDX node used to facilitate interfacing signaling between a digital domain and an analog domain of a FDX or other communication network.