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.

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 one or more beams to acquire sound from one or more specific locations in the predetermined area.

Abstract: A circuit includes a digital-to-analog conversion circuit, an amplifying circuit, a mixing circuit, and an analog-to-digital converter. The digital-to-analog conversion circuit is configured to receive and convert audio data in the digital domain and output audio data in the analog domain. The audio data includes a main-tone component. The amplifying circuit is configured to output an audio signal according to the audio data in the analog domain. The mixing circuit is configured to eliminate the main-tone component according to the audio data in the analog domain and the audio signal and to output a feedback signal. The analog-to-digital converter is configured to convert the feedback signal from the analog domain to the digital domain.

Abstract: What is described is a method for associating noises of at least one signal class (like disturbing noise) of a plurality of signal classes (like disturbing noise and non-disturbing noise). The method has the steps of “receiving environmental noises” and “establishing whether the environmental noises or a set of parameters derived from the environmental noises fulfill/s a predefined rule which describes the signal class of the plurality of signal classes”. Starting from this, the steps of “logging that the predefined rule has been fulfilled”, “recording the environmental noises received for a migrating time window”, “deriving a set of parameters from the environmental noises for the migrating time window and storing the set of parameters” or “emitting an activation signal for another device for associating a noise” are performed.

Abstract: A controller for an acoustic echo canceller includes a noise estimator configured to estimate a level of noise that is comprised in a microphone signal relative to an echo component, estimated by the acoustic echo canceller, comprised in the microphone signal. The controller further includes a control module configured to control the acoustic echo canceller in dependence on that estimate.

Abstract: A signal processing method performs echo reduction processing on at least one of a collected sound signal of a first microphone, a collected sound signal of a second microphone, or both the collected sound signal of the first microphone and the collected sound signal of the second microphone, and calculates a correlated component between the collected sound signal of the first microphone and the collected sound signal of the second microphone, using a collected sound signal of which echo has been reduced by the an echo reduction processing.

Abstract: A signal processing method performs echo reduction processing on at least one of a collected sound signal of a first microphone, a collected sound signal of a second microphone, or both the collected sound signal of the first microphone and the collected sound signal of the second microphone, and calculates a correlated component between the collected sound signal of the first microphone and the collected sound signal of the second microphone, using a collected sound signal of which echo has been reduced by the an echo reduction processing.

Abstract: According to an embodiment, a noise cancellation method comprises updating a first far-end channel filter according to a first variance yielded based on an input signal and a first far-end signal included in the input signal, providing a first output signal obtained by removing a first far-end noise signal from the input signal through the first far-end channel filter, updating a second far-end channel filter according to a second variance yielded based on the first output signal and a second far-end signal included in the input signal, and providing a second output signal obtained by removing a second far-end noise signal from the first output signal through the second far-end channel filter.

Abstract: A device-management system performs audio processing, such as acoustic echo cancellation or beamforming, in a computing-resource allocation corresponding to a functionally limited device. The device-management system may be a locally-connected network device that is in communication with one or more user devices; the device-management system may also or instead be a remote device that communicates with the user devices using the locally-connected network device. The device-management system may receive audio data from one or more microphones of one or more user devices. To perform acoustic echo cancellation, the device-management system may receive and process time data corresponding to a time of output of audio by the user device.

Abstract: The technology described herein can be embodied in a method for estimating a power spectral density of noise, the method including receiving an input signal representing audio captured using a microphone. The input signal includes a first portion that represents acoustic outputs from two or more audio sources, and a second portion that represents a noise component. The method also includes iteratively modifying a frequency domain representation of the input signal, such that the modified frequency domain representation represents a portion of the input signal in which effects due to the first portion are substantially reduced. The method further includes determining, from the modified frequency domain representation, an estimate of a power spectral density of the noise, and generating a control signal configured to adjust one or more gains of an acoustic transducer. The control signal is generated based on the estimate of the power spectral density of the noise.

Abstract: An echo cancellation detector for controlling an acoustic echo canceller that is configured to cancel an echo of a far-end signal in a near-end signal in a telephony system, the echo cancellation detector comprising a comparison generator configured to compare the far-end signal with the near-end signal, a decision unit configured to make a determination about a first acoustic echo canceller based on that comparison and a controller configured to control an operation of a second acoustic echo canceller in dependence on the determination.

Abstract: A method for dynamically modifying a frame structure in transceiving of data between a network side entity and at least one node, the data is divided among a plurality of frames, the method comprising: providing an upstream management symbol offset position with respect to a frame reference point, the upstream management symbol offset position is associated with an upstream management symbol that is configured to at least convey downstream acknowledgement data; providing an updated frame structure to the at least one node, including: information defining a transition point between downstream and upstream; and a reference point indicating when the updated frame structure shall take effect; receiving the updated frame structure by the at least one node; and employing the updated frame structure from the reference point, by the network side entity.

Abstract: The disclosure describes embodiments for reducing adjacent channel interference for wireless vehicular messages. In some embodiments, a method includes monitoring, by an onboard vehicle computer, a channel of a Vehicle-to-Everything (V2X) radio for a V2X message. The method includes determining waveform data describing a waveform that is measured on the channel by the onboard vehicle computer when monitoring the channel. The method includes determining interference data which describes an adjacent channel interference waveform which is measured on the channel by the onboard vehicle computer when monitoring the channel. The waveform includes the adjacent channel interference waveform.

Abstract: A noise elimination device includes a first converter, a second converter, and a signal processing unit. The first converter is configured to convert an input signal input from a microphone into digital data to generate input digital data. The second converter is configured to convert an internal noise signal into digital data. The signal processing unit is configured to use, as reference data, digital data corresponding to an output signal output to a speaker and digital data having been converted by the second converter, and eliminate a component corresponding to the reference data from the input digital data.

Abstract: A computing device is provided that may include a sound processing device configured to output sound output frames via a sound output device and capture sound input frames via a sound input device. The computing device further includes a processor configured to render a stream of sound output frames based on at least sound output data received from a remote computing device over a computer network, provide reference frames for each of the sound output frames to a reference frame buffer of an acoustic echo cancellation module. The processor may be further configured to detect a sound output frame rendering stall condition, and based on detecting the sound output frame rendering stall condition, suppress the acoustic echo cancellation module and send the stream of sound input frames to the remote computing device without being processed by the acoustic echo cancellation module.

Abstract: The method includes inputting first groups of frequency samples to be transmitted over, respectively received from, first and second subsets of subscriber lines at respective first tones and following a first group selection order, alternated with mirrored second groups of frequency samples to be transmitted over, respectively received from, the second subset of subscriber lines at respective mirrored second tones of the first tones, and following a mirrored second group selection order of the first group selection order, to a vectoring processor for joint mitigation of first crosstalk present over the first and second subsets of subscriber lines at the respective first tones and second crosstalk present over the second subset of subscriber lines at the respective second tones. The first tones are common to the first and second transmit spectra. Mirroring is with respect to a folding frequency derived from a reference sampling frequency applicable to the first subset of subscriber lines.

Abstract: A gain control system for dynamically tuning an echo canceller, the echo canceller being configured to estimate an echo of a far-end signal and subtract that echo estimate from a microphone signal to output an echo cancelled signal, the gain control system comprising a monitoring unit configured to estimate an energy associated with an impulse response of an adaptive filter configured to generate the echo estimate from the far-end signal and a gain tuner configured to adjust an attenuation of at least one of the microphone signal and the far-end signal in dependence on the estimated energy.

Abstract: A method of controlling an electronic device having a battery and a speaker is disclosed. The method includes receiving an audio data signal, monitoring a battery-related parameter of the battery, determining whether the monitored battery-related parameter traverses a threshold, and bit-shifting the audio data signal based on the monitored battery-related parameter traversing the threshold. Also disclosed is an attenuator circuit for an electronic device having a battery.

Abstract: The disclosure describes embodiments for reducing adjacent channel interference for wireless vehicular messages. In some embodiments, a method includes monitoring, by an onboard vehicle computer, a channel of a Vehicle-to-Everything (V2X) radio for a V2X message. The method includes determining waveform data describing a waveform that is measured on the channel by the onboard vehicle computer when monitoring the channel. The method includes determining interference data which describes an adjacent channel interference waveform which is measured on the channel by the onboard vehicle computer when monitoring the channel. The waveform includes the adjacent channel interference waveform.

Abstract: An audio processing method including: generating a plurality of frequency spectra by transforming a plurality of audio signals inputted to a plurality of input devices respectively, comparing an amplitude of each of frequency components of a specific frequency spectrum included in the plurality of frequency spectra with an amplitude of each of frequency components of one or a more other frequency spectra different from the specific frequency spectrum included in the plurality of frequency spectra, for each of the frequency components, extracting, from the frequency components, a frequency component in which an amplitude of the specific frequency spectrum is larger than an amplitude of the one or more other frequency spectra, and controlling an output corresponding to the plurality of audio signal inputted to each of the plurality of input devices based on a proportion of the extracted frequency component in the frequency components whose amplitudes has been compared.

Abstract: A resource scheduling method and apparatus in a DSL system, and a system are provided, to reduce energy consumption of the DSL system. The method includes: determining, according to service traffic information of each subscriber line and transmission capability information of each subscriber line, a transmission opportunity initial value allocated to each subscriber line in a TDD frame; and according to the transmission opportunity initial value of each subscriber line, dividing the TDD frame into a normal operation interval and a discontinuous operation interval, determining a transmission opportunity occupied by each subscriber line in the normal operation interval, and determining a subscriber line group in the discontinuous operation interval and a transmission opportunity occupied by each subscriber line group in the discontinuous operation interval.

Abstract: The technology described herein can be embodied in a method for estimating a power spectral density of noise, the method including receiving an input signal representing audio captured using a microphone. The input signal includes a first portion that represents acoustic outputs from two or more audio sources, and a second portion that represents a noise component. The method also includes iteratively modifying a frequency domain representation of the input signal, such that the modified frequency domain representation represents a portion of the input signal in which effects due to the first portion are substantially reduced. The method further includes determining, from the modified frequency domain representation, an estimate of a power spectral density of the noise, and generating a control signal configured to adjust one or more gains of an acoustic transducer. The control signal is generated based on the estimate of the power spectral density of the noise.

Abstract: Systems and methods are provided for estimating non-linearity of a transmitter, a receiver or both based on measurements and/or feedback. The estimation of the non-linearity may be used at the transmitter, the receiver or both to tune one or more components to reduce non-linearity. In one aspect, a method for wireless communications comprises generating at least one training signal, and outputting the at least one training signal for transmission to a wireless node. The method also comprises receiving a feedback message from the wireless node, the feedback message providing feedback of the at least one training signal received at the wireless node. The method further comprises tuning at least one component based on the feedback message.

Abstract: The present disclosure relates to near-end crosstalk (NEXT) cancellation. A transmit communication signal is transmitted over a first Digital Subscriber Line (DSL) connection using a first group of frequencies and a receive communication signal is received over a second DSL connection using a second group of frequencies that at least partially overlaps the first group of frequencies. A crosstalk correlation between the first and second communication signals is determined. Based on the crosstalk correlation, a crosstalk cancellation signal is generated. The crosstalk cancellation signal is subtracted from the second communication signal, with the intention of reducing NEXT.

Abstract: A processor of an apparatus selects a codebook from a plurality of codebooks embedded in a quasi-cyclic-low-density parity-check (QC-LDPC) code. The processor stores the selected codebook in a memory associated with the processor. The processor also encodes data using the selected codebook to generate a plurality of modulation symbols of the data. The processor further controls a transmitter of the apparatus to multiplex, convert, filter, amplify and radiate the modulation symbols as electromagnetic waves through one or more antennas of the apparatus. In selecting the codebook from the plurality of codebooks embedded in the QC-LDPC code, the processor selects the codebook according to one or more rules such that a small codebook requiring a shorter amount of processing latency for the encoding is selected for the encoding unless a larger codebook corresponding to a larger amount of processing latency for the encoding is necessary for the encoding.

Abstract: A two-way conversation assisting device includes a first microphone that enters a first voice, a first loudspeaker that outputs the first voice, a second microphone that enters a second voice, a second loudspeaker that outputs the second voice, and a first echo and crosstalk canceller. The first echo and crosstalk canceller estimates and calculates, using an input signal into the second loudspeaker, a first interference signal indicative of degrees of a first echo caused when the second voice output from the second loudspeaker enters into the first microphone and first crosstalk caused when the second voice enters into the first microphone, and removes the calculated first interference signal from an output signal of the first microphone.

Abstract: Aspects of the present disclosure include cancellation systems, methods, apparatuses, and software for acoustic or electromagnetic measurement or communications. In some embodiments, full-duplex measurement and/or communications systems can be implemented that permit simultaneous transmission and reception of signals. A cancellation signal generated in accordance with teachings herein may be produced by iteratively solving a Wiener filter problem and provided to a cancellation transducer to cancel unwanted signals in an analog fashion and/or adaptive cancellation module to cancel unwanted signals in a digital fashion. A number of new techniques for reducing or eliminating unwanted signals are described herein that can be used alone or in combination in order to tailor a signal cancelling system, method, apparatus, or software product to particular uses and environments.

Abstract: A processor of an apparatus establishes a wireless communication link with at least one other apparatus via a transceiver of the apparatus. The processor wirelessly communicates with the other apparatus via the wireless communication link by: selecting a first shift-coefficient table from a plurality of shift-coefficient tables; generating a QC-LDPC code using a base matrix and at least a portion of the first shift-coefficient table; selecting a codebook from a plurality of codebooks embedded in the QC-LDPC code; storing the selected codebook in a memory associated with the processor; encoding data using the selected codebook to generate a plurality of modulation symbols of the data; and controlling the transceiver to multiplex, convert, filter, amplify and radiate the modulation symbols as electromagnetic waves through one or more antennas of the apparatus to transmit the modulation symbols of the data to the other apparatus via the wireless communication link.

Abstract: A processor of an apparatus establishes a wireless communication link with at least one other apparatus via a transceiver of the apparatus. The processor wirelessly communicates with the other apparatus via the wireless communication link by: selecting a first shift-coefficient table from a plurality of shift-coefficient tables; generating a QC-LDPC code using a base matrix and at least a portion of the first shift-coefficient table; selecting a codebook from a plurality of codebooks embedded in the QC-LDPC code; storing the selected codebook in a memory associated with the processor; encoding data using the selected codebook to generate a plurality of modulation symbols of the data; and controlling the transceiver to multiplex, convert, filter, amplify and radiate the modulation symbols as electromagnetic waves through one or more antennas of the apparatus to transmit the modulation symbols of the data to the other apparatus via the wireless communication link.

Abstract: A double-talk detecting method including estimating a noise contained in a mixed signal containing at least an echo, a near-end signal and a noise to determine an estimated noise; and detecting a presence of a near-end signal contained in said mixed signal using said estimated noise.

Abstract: Example techniques involve noise-robust acoustic echo cancellation. An example implementation may involve causing one or more speakers of the playback device to play back audio content and while the audio content is playing back, capturing, via the one or more microphones, audio within an acoustic environment that includes the audio playback. The example implementation may involve determining measured and reference signals in the STFT domain. During each nth iteration of an acoustic echo canceller (AEC): the implementation may involve determining a frame of an output signal by generating a frame of a model signal by passing a frame of the reference signal through an instance of an adaptive filter and then redacting the nth frame of the model signal from an nth frame of the measured signal. The implementation may further involve determining an instance of the adaptive filter for a next iteration of the AEC.

Abstract: A method of cancelling noise present in a data signal received on an electrical bifilar line, is implemented by a sender-receiver device comprising a first transformer, termed the differential mode circuit, the primary side of the first transformer connected by two wires to the bifilar line, a second transformer, termed the common mode circuit, the primary side of the second transformer connected to the primary side of the differential mode circuit, and to ground. The method comprises during an adjustment phase: obtaining a first voltage value on the bifilar line, termed the differential mode voltage; obtaining a second voltage value at the common mode circuit, termed the image voltage, resulting from the differential mode voltage; calculating the ratio between the second and first voltage values, termed the conversion ratio.

Abstract: Provided is a method and computer program product for producing an enhanced audio signal for an output device from audio signals received by 2 or more microphones in close proximity to each other. For example, one embodiment of the present invention comprises the steps of receiving a first input audio signal from the first microphone, digitizing the first input audio signal to produce a first digitized audio input signal, receiving a second input audio input signal from the second microphone, digitizing the second input audio input signal to produce a second digitized audio input signal, using the first digitized audio input signal as a reference signal to an adaptive prediction filter, using the second digitized audio input signal as input to said adaptive prediction filter and finally adding a prediction result signal from the adaptive prediction filter to the first digitized audio input signal to produce the enhanced audio signal.

Abstract: A method includes obtaining, by a processor, an audio echo signal and an audio desired signal from an acoustic echo correction stage of an electronic device, and converting the echo signal and the desired signal to the frequency domain. The method further includes grouping, by the processor, frequency bin results of respective frequency domain converted echo and desired signals into respective echo and desired sub-bands. A sub-band suppressor gain is estimated based on an estimated sub-band energy for the echo and desired sub-bands. The method further includes modulating the frequency domain converted desired signal to compensate for residual echo, the modulating based, at least in part, on the estimated sub-band suppressor gain, and the modulating producing a compensated frequency domain converted echo signal. The method also includes converting the compensated frequency domain converted desired signal into time domain converted audio output signal.

Abstract: A sound source separation device includes a first microphone that picks up a first voice, a second microphone that picks up a second voice, a first crosstalk canceller that removes, from a voice signal of the first microphone, first crosstalk caused when the second voice is picked up by the first microphone, and a second crosstalk canceller that removes, from a voice signal of the second microphone, second crosstalk caused when the first voice is picked up by the second microphone. The first crosstalk canceller uses a voice signal in which the second crosstalk is removed from the voice signal of the second microphone to estimate and calculate a first interference signal indicative of a degree of the first crosstalk, and to remove the calculated first interference signal from the voice signal of the first microphone.

Abstract: A method and system for improving an echo in a hands-free call of a mobile terminal may include the mobile terminal comprising a speaker, a primary microphone and an auxiliary microphone; the distance from the primary microphone to the speaker can be less than that from the auxiliary microphone to the speaker; and the method may include using the auxiliary microphone to acquire outside sound during a hands-free call.

Abstract: The present disclosure discloses an echo cancellation method and apparatus. The method includes: obtaining a far-end signal and a near-end signal; detecting whether a speech signal exists in the near-end signal; processing the far-end signal according to a normalized least mean square adaptive filtering (NLMS) algorithm to obtain an estimated signal, and subtracting the estimated signal from the near-end signal to obtain a residual signal; after it is detected that the speech signal exists in the near-end signal, detecting whether the residual signal meets a preset output condition; and when it is detected that the residual signal does not meet the preset output condition, multiplying the residual signal by a first attenuation factor to obtain an output signal.

Abstract: A method of active noise reduction. The method comprises instructing a microphone electronically coupled by a client terminal to record a nonaural noise signal, instructing a circuit of the client terminal to record an aural noise signal using at least one electroacoustic transducer of an earphone, calculating a noise reduction signal based on a function combining nonaural noise signal and the aural noise signal, calculating a noise reduced signal based on a combination of a content signal prepared to be played by the at least one electroacoustic transducer and the noise reduction signal, and instructing the circuit to play the noise reduced signal via the at least one electroacoustic transducer. The nonaural noise signal and the aural noise signal are recorded at least partly simultaneously.

Abstract: Disclosed herein is a method of enhancing speech. The method includes calculating a far-end speech spectrum by performing fast Fourier transformation of a signal received by a far-end user, calculating a background noise spectrum collected by a microphone provided to a mobile device of a near-end user; calculating a gain from the far-end speech spectrum and the background noise spectrum using a speech intelligibility index-based module, and deriving an enhanced far-end speech spectrum by applying the gain to the far-end speech spectrum, wherein, in calculating a gain using a speech intelligibility index-based module, a power budget used for transmitting and receiving a speech signal is set to vary with the background noise spectrum.

Abstract: Embodiments described herein provide echo cancellation power saving management at a cable transceiver. An echo response signal having a first number of signal components is obtained, via an echo cancellation filter. At a first iteration for calculating a first accumulative echo power, a respective echo tap that corresponds to the first iteration is identified. The first accumulative echo power is calculated for the respective iteration by summing powers of outputs from a last echo tap to the respective echo tap. It is then determined whether the first accumulative echo power, exceeds a pre-determined echo power threshold. If the first accumulative echo power exceeds the pre-determined echo power threshold, a first turn-off indication is sent to the echo cancellation filter to turn off all echo taps including and between the last echo tap to the first echo tap.

Abstract: Provided are, among other things, systems, methods and techniques for reducing echo in an audio signal. One representative embodiment involves obtaining an input signal, an estimate of a system-characterizing function, and a reference signal, each at a corresponding sample rate and each divided into a plurality of sub-bands; separately processing such sub-bands, where for a given sub-band the estimate of the system-characterizing function and the reference signal are processed to generate an echo-estimation signal and then the echo-estimation signal is subtracted from the input signal to provide an echo-corrected signal for such given sub-band; and combining the echo-corrected signal from each of different ones of the plurality of the sub-bands to provide a final output signal, with the echo-estimation signal generated using a processing sample rate that is lower than the sample rate for the input signal.

Abstract: An acoustic echo canceller includes an adaptive filter and a double-talk detector. The adaptive filter includes a linear filter and a coefficient calculator. The linear filter has a transfer function that is controlled by a set of variable filter coefficients and that is configured to cancel an estimate of echo in a microphone signal to provide an output signal. The coefficient calculator is configured to update the set of variable filter coefficients based on a variable adaptation rate. The double-talk detector is configured to calculate changes in the energy of the variable filter coefficients (between updates of the coefficients). The acoustic echo canceller is configured to adjust the variable adaptation rate based on whether the energy of the variable filter coefficients is determined to be either oscillating or steadily changing (increasing or decreasing).

Abstract: There is provided methods and apparatuses for decoding and encoding of audio signals. In particular, a method for decoding includes receiving a waveform-coded signal having a spectral content corresponding to a subset of the frequency range above a cross-over frequency. The waveform-coded signal is interleaved with a parametric high frequency reconstruction of the audio signal above the cross-over frequency. In this way an improved reconstruction of the high frequency bands of the audio signal is achieved.

Abstract: Techniques of performing acoustic echo cancellation involve providing a bi-magnitude filtering operation that performs a first filtering operation when a magnitude of an incoming audio signal to be output from a loudspeaker is less than a specified threshold and a second filtering operation when the magnitude of the incoming audio signal is greater than the threshold. The first filtering operation may take the form of a convolution between the incoming audio signal and a first impulse response function. The second filtering operation may take the form of a convolution between a nonlinear function of the incoming audio signal and a second impulse response function. For such a convolution, the bi-magnitude filtering operation involves providing, as the incoming audio signal, samples of the incoming audio signal over a specified window of time. The first and second impulse response functions may be determined from an input signal input into a microphone.

Abstract: One or more embodiments set forth an audio presence detection system, with a memory that includes an acoustic presence detection application, and a processor that executes the acoustic presence detection application. The audio presence detection system receives a first input signal associated with a first speaker, and receives a second input signal associated with a first microphone. The audio presence detection system generates, via an adaptive filter, a first estimation signal based on at least the first input signal and an impulse response associated with a room related to the first speaker and the first microphone. The audio presence detection system computes a first error signal based on the second input signal and the first estimation signal. The audio presence detection system determines that an object is present within a space based on a magnitude associated with the first error signal relative to a first threshold level.

Abstract: A method and system for assisting communication. The method includes receiving a message in a first communication medium from a first device of a first user. The message has one or more recipients. The method further includes determining whether at least one recipient of the one or more recipients is in a predefined category by referencing a stored look-up table, monitoring, via one or more sensors, surroundings of the at least one recipient when the at least one recipient is in the predefined category, detecting whether there is a hazard based on the monitoring, and determining whether the first communication medium corresponds to a preferable communication medium based on the predefined category of the at least one recipient. Further, the method includes converting the message to the preferable communication medium and delivering the converted message to at least one recipient device when a hazard is not detected.

Abstract: An acoustic echo cancellation (AEC) system that detects and compensates for differences in delay times between the AEC system and a set of wireless speakers. The filter coefficients used for AEC are adjusted based on the determined delay time to correct for frequency domain signal rotation.

Abstract: A system, article, and method of audio quality and latency adjustment for audio processing by using audio feedback.