Abstract: An apparatus and/or method discloses a video conference with enhanced audio quality using high-fidelity audio sharing (“HAS”). In one embodiment, a network connection between a first user equipment (“UE”) and a second UE is established via a communication network for providing an interactive real-time meeting. After sending a first calibration audio signal from the first UE to the second UE, a second calibration audio signal is retuned from the second UE to the first UE according to the first calibration audio signal. Upon identifying a far end audio (“FEA”) delay based on the first calibration audio signal and the second calibration audio signal, a first mixed audio data containing the first shared audio data and first FEA data is fetched from an audio buffer. The first FEA data is subsequently removed or extracted from the mixed audio data in response to the FEA delay.

Abstract: A method for separating sound sources includes digitizing acoustic signals from a plurality of sources with a plurality of microphone arrays, wherein each of microphone arrays includes one or more microphones, wherein at least one of the microphone arrays may be asynchronous to another one of the microphone arrays or may be moving with respect to another one of the microphone arrays. Spatial parameters are estimated of the digitized acoustic signals. The method includes estimating time-varying source spectra for the sources from the digitized acoustic signals as a function of the digitized acoustic signals and data received from at least one other microphone array. Source signals are estimated for one or more of the sources by filtering the digitized acoustic data digitized at the respective microphone array using the spatial parameters of the digitized acoustic data and the time-varying source spectra from all or a subset of the microphone arrays.

Abstract: A double-talk state detection method includes: calculating an energy ratio between a first energy of an error signal in each sub-band of M sub-bands and a second energy of a filtered signal in the same sub-band as the error signal, thereby obtaining M energy ratios, where the error signal is a difference between an input signal collected by a microphone and the filtered signal, the filtered signal is a signal obtained after performing filtering process on a reference signal, and M is a positive integer; performing a first smoothing processing on the M energy ratios to obtain M first energy smoothing ratios, and performing a second smoothing processing on the M first energy smoothing ratios to obtain M second energy smoothing ratios; performing double-talk state detection based on the M first energy smoothing ratios and the M second energy smoothing ratios to determine a state of the input signal.

Abstract: A signal processing apparatus includes: a coefficient update unit configured to approximate a characteristic of a transmission line of an optical signal by a first tap coefficient vector of which an L0 norm is a predetermined value or less; a zeroing unit configured to generate a second tap coefficient vector by replacing, with 0, a tap coefficient of which an absolute value is less than a threshold among tap coefficients of the first tap coefficient vector; and an adaptive filter configured to perform, based on the second tap coefficient vector, adaptive equalization processing on a digital signal corresponding to an optical signal received via the transmission line.

Abstract: A method, computer program, and computer system is provided for an all-deep-learning based AEC system by recurrent neural networks. The model consists of two stages, echo estimation stage and echo suppression stage, respectively. Two different schemes for echo estimation are presented herein: linear echo estimation by multi-tap filtering on far-end reference signal and non-linear echo estimation by single-tap masking on microphone signal. A microphone signal waveform and a far-end reference signal waveform are received. An echo signal waveform is estimated based on the microphone signal waveform and a far-end reference signal waveform. A near-end speech signal waveform is output based on subtracting the estimated echo signal waveform from the microphone signal waveform, and echoes are suppressed within the near-end speech signal waveform.

Abstract: A method is disclosed, the method comprising obtaining at least one first information indicative of audio data gathered by at least one first microphone, and at least one second information indicative of audio data gathered by at least one second microphone; determining a differential information indicative of one or more differences between at least two pieces of information, wherein the differential information is determined based, at least in part, on the at least one first information and the at least one second information; and compensating of an impact onto the audio data, wherein audio data of the first information and/or the second information is compensated based, at least in part, on the determined differential information. Further, an apparatus, and a system are disclosed.

Abstract: A method includes receiving a microphone audio signal and a playout audio signal, and determining a frequency representation of the microphone audio signal and a frequency representation of the playout audio signal. For each frequency representation, the method also includes determining features based on the frequency representation. Each feature corresponds to a pair of frequencies of the frequency representation and a period of time between the pair of frequencies. The method also includes determining that a match occurs between a first feature based on the frequency representation of the microphone audio signal and a second feature based on the frequency representation of the playout audio signal, and determining that a delay value between the first feature and the second feature corresponds to an echo within the microphone audio signal.

Abstract: A method for receiving audio broadcasts during a digital enhanced cordless telecommunication (DECT) frame, the method may include (a) opening, during a DECT frame and by a portable point (PP) that is a potential recipient of the audio broadcasts from a fixed part (FP), multiple reception windows that are spaced apart from each other and are allocated to receive signals related to audio broadcast; (b) receiving received signals, by the PP and during the multiple spaced apart reception windows; and (c) processing, by the PP, the received signals.

Abstract: Audio systems and methods are provided that receive a sound management input signal as a reference signal to remove related content from a microphone signal in, e.g., an automotive hands-free system. The sound management signal may provide an acoustic augmentation to reduce, enhance, or create an acoustic effect, e.g., of an engine, motor, or other operating components. A signal processor receives the sound management signal and the microphone signal, and reduces or removes the sound management signal components from the microphone signal.

Abstract: Echo reduction. At least one example embodiment is a method including producing, by a loudspeaker, acoustic waves based on a far-microphone signal; receiving, at a local microphone, an echo based on the acoustic waves, and receiving acoustic waves generated locally, the receiving creates a local-microphone signal; producing an estimated-echo signal based on the far-microphone signal and a current step-size parameter; summing the local-microphone signal and the estimated echo signal to produce a resultant signal having reduced echo in relation to the local-microphone signal; and controlling the current step-size parameter. The controlling current step size may include: calculating a convergence value based on a cross-correlation of the local-microphone signal and the resultant signal; and updating the current step-size parameter based on the convergence value.

Abstract: PTP timestamps are obtained and correlated to internal audio time and timestamps. This allows Ethernet/IP audio devices to be closely aligned with local digital audio devices to allow improved AEC operations. By performing repetitive PTP timestamps based on local audio time, drift between PTP time and local audio time is determined and corrected. By performing the PTP to local audio time operations on each independent Ethernet network, the Ethernet/IP audio devices can be split between independent LANs. If some local digital audio inputs formed from analog inputs are not aligned with other local digital audio inputs, signals can be provided in analog format and audio samples of the signals can be correlated to local audio time. With correlations performed as necessary based on the various audio inputs in use, high quality AEC operations are performed.

Abstract: A mapping method for identifying density which includes determining at least one geographical boundary about a geographical location of interest, the at least one geographical boundary is free of any self-intersections, dividing the at least one geographical boundary into a plurality of regions of interests, each region of interest is defined by a start point and an end point on the at least one geographical boundary, wherein the end point associated with one region of interest coincides with a start point of a neighboring region of interest wherein the region of interest falls within the at least one geographical boundary, for each of the plurality of regions of interest, calculating at least one parameter of interest within the region of interest, and graphically presenting a segment between the start point and the end point on the at least one geographical boundary with a line thickness proportional to the calculation results.

Abstract: A method (400) includes receiving a microphone audio signal (132) and a playout audio signal (112), and determining a frequency representation (324) of the microphone audio signal and a frequency representation of the playout audio signal. For each frequency representation, the method also includes determining features (302) based on the frequency representation. Each feature corresponds to a pair of frequencies (342) of the frequency representation and a period of time between the pair of frequencies. The method also includes determining that a match (212) occurs between a first feature based on the frequency representation of the microphone audio signal and a second feature based on the frequency representation of the playout audio signal, and determining that a delay value (222) between the first feature and the second feature corresponds to an echo within the microphone audio signal.

Abstract: An apparatus of reducing feedback noise in an acoustic system, the apparatus comprising: a first input for receiving a first signal derived from a first microphone associated with a first channel, the first signal comprising a first set of frequency sub-bands; a second input for receiving a second signal derived from a second microphone associated with a second channel, the second signal comprising second set of frequency sub-bands, the first and second sets of frequency sub-bands having matching frequency ranges, each frequency sub-band of the first and second sets of frequency sub-bands having a frequency of greater than a threshold frequency; and one or more processors configured to: determining feedback at a first speaker associated with the first channel; and responsive to determining feedback, mix each of the first set of frequency sub-bands with a corresponding one of the second set of frequency sub-bands to generate a mixed output signal comprising a mixed set of frequency sub-bands; wherein the mixin

Abstract: The present invention provides an interference cancellation circuit, wherein the interference cancellation circuit includes a PAPR detection circuit, a control circuit and a filter. In the operations of the interference cancellation circuit, the PAPR detection circuit is configured to detect a PAPR of a signal in a spectrum in a real-time manner to generate a detection result. The control circuit is configured to generate a control signal according to the detection result. The filter is configured to determine a filtering frequency point of the filter according to the control signal, and to filter the signal to generate an output signal.

Abstract: Systems and methods for improved acoustic echo cancellation are provided. In various embodiments, a microphone located in the loudspeaker enclosure provides a first signal that is used to estimate the loudspeaker displacement which is proportional to the sound pressure level (SPL) inside the enclosure. A second signal is then derived by mapping the displacement to the loudspeaker's force factor (Bl(x)) and then modulating this by a measured current to a voice coil inside the speaker to provide an estimate of the force acting on the moving mass of the loudspeaker. The first signal is highly correlated with the echo signal for low frequencies and the second signal is highly correlated with the echo signal for high frequencies. The two signals are then combined to provide a single improved AEC reference signal.

Abstract: Techniques for detecting a voice command from a user of a hearable device. The hearable device may include an in-ear facing microphone to capture sound emitted from an ear of the user, and an exterior facing microphone to capture sound emitted from an exterior environment of the user. The in-ear microphone may generate an in-ear audio signal representing the sound emitted from the ear, and the exterior microphone may generate an exterior audio signal representing sound from the exterior environment. The hearable device may include components to determine correlations or similarities between the in-ear audio signal and exterior audio signal, which indicate that the audio signals represent sound emitted from the user. Further, the components may perform voice activity detection to determine that the sound emitted from the user is a voice command, and proceed to perform further voice-processing techniques.

Abstract: A software-based conferencing platform is provided. The platform comprises a plurality of audio sources providing input audio signals, the audio sources including a virtual audio device driver configured to receive far-end input audio signals from a conferencing software module, and a network audio library configured to receive near-end input audio signals from one or more near-end audio devices. The platform further comprises a digital signal processing component configured to receive the input audio signals from the audio sources and generate audio output signals based the received signals, the digital signal processing component comprising an acoustic echo cancellation module configured to apply acoustic echo cancellation techniques to one or more of the near-end input audio signals.

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

Abstract: In an audio processing system (300), a filtering section (350, 400): receives subband signals (410, 420, 430) corresponding to audio content of a reference signal (301) in respective frequency subbands; receives subband signals (411, 421, 431) corresponding to audio content of a response signal (304) in the respective subbands; and forms filtered inband references (412, 422, 432) by applying respective filters (413, 423, 433) to the subband signals of the reference signal.

Abstract: A method for managing network congestion is provided. The method comprises: receiving, at a receiver, a packet comprising a timestamp provided by a first clock of a sender; deriving, by the receiver, a latency value based at least in part on the timestamp provided by the first clock and a receipt time provided by a second clock of the receiver; determining a latency change by comparing the latency value with a previous latency value; and determining a state of network congestion based at least in part on the latency change.

Abstract: A method, for preventing the intelligible voice recording is provided. The voice of a subject or interlocutor is recorded for a given time interval thereby providing a voice recording. The voice recording is cut into shorter time interval segments thereby providing a set of voice recording segments. The set of voice recording segments is mixed in a randomly rearranged order. The mixed set of voice recording segments is spliced into a single randomly mixed voice recording. Emitting the randomly mixed voice recording during speaking of the subject or interlocutor prevents the intelligible recording of the voice of the subject or interlocutor.

Abstract: A digital circuit for implementing a channel-tracking functionality, in which an adaptive (e.g., FIR) filter is updated based on reinforcement learning. In an example embodiment, the adaptive filter may be updated using an LMS-type algorithm. The digital circuit may also include an electronic controller configured to change the convergence coefficient of the LMS algorithm using a selection policy learned by applying a reinforcement-learning technique and based on residual errors and channel estimates received over a sequence of iterations. In some embodiments, the electronic controller may include an artificial neural network. An example embodiment of the digital circuit is advantageously capable of providing improved performance after the learning phase, e.g., for communication channels exhibiting variable dynamicity patterns, such as those associated with aerial copper cables or some wireless channels.

Abstract: A system configured to perform echo cancellation using a reduced number of reference signals. The system may perform multi-channel acoustic echo cancellation (MCAEC) processing on a first portion of a microphone audio signal that corresponds to early reflections and may perform single-channel acoustic echo cancellation (AEC) processing on a second portion of the microphone audio signal that corresponds to late reverberations. For example, the system may use MCAEC processing on a plurality of reference audio signals to generate a first echo estimate signal and may subtract the first echo estimate signal from the microphone audio signal to generate a residual audio signal. The system may delay the first echo estimate signal, perform the AEC processing to generate a second echo estimate signal, and subtract the second echo estimate signal from the residual audio signal to generate an output audio signal. This reduces an overall complexity associated with performing echo cancellation.

Abstract: Techniques to dynamically engage echo cancellation are described. In one embodiment, an apparatus may comprise a streaming component operative to establish a audio connection between the first client device and a second client device via the network interface controller; and receive a far-end audio stream at the first client device from the second client device via the audio connection; an audio capture component operative to capture a near-end audio stream at the first client device; and an echo processing component operative to compare the near-end audio stream and the far-end audio stream to determine whether a far-end echo is present in the near-end audio stream; and use an echo-cancellation module at the first client device where the far-end echo is present in the near-end audio stream. Other embodiments are described and claimed.

Abstract: A digital circuit for implementing a channel-tracking functionality, in which an adaptive (e.g., FIR) filter is updated based on reinforcement learning. In an example embodiment, the adaptive filter may be updated using an LMS-type algorithm. The digital circuit may also include an electronic controller configured to change the convergence coefficient of the LMS algorithm using a selection policy learned by applying a reinforcement-learning technique and based on residual errors and channel estimates received over a sequence of iterations. In some embodiments, the electronic controller may include an artificial neural network. An example embodiment of the digital circuit is advantageously capable of providing improved performance after the learning phase, e.g., for communication channels exhibiting variable dynamicity patterns, such as those associated with aerial copper cables or some wireless channels.

Abstract: Systems and methods for processing an audio signal are provided for replay on an audio device. An audio signal is spectrally decomposed into a plurality of subband signals using band pass filters. Each of the subband signals are provided to a respective modulator and subsequently, from the modulator output, provided to a respective first processing path that includes a first dynamic range compressor, DRC. Each subband signal is feedforward compressed by the respective first DRC to obtain a feedforward-compressed subband signal, wherein the first DRC is slowed relative to an instantaneous DRC. Subsequently, each feedforward-compressed subband signal is provided to a second processing path that includes a second DRC, wherein the feedforward-compressed subband signal is compressed by the respective second DRC and outputted to the respective modulator. Modulation of the subband signals is then performed in dependence on the output of the second processing path.

Abstract: An apparatus and/or method discloses a video conference with enhanced audio quality using high-fidelity audio sharing (“HAS”). In one embodiment, a network connection between a first user equipment (“UE”) and a second UE is established via a communication network for providing an interactive real-time meeting. After sending a first calibration audio signal from the first UE to the second UE, a second calibration audio signal is retuned from the second UE to the first UE according to the first calibration audio signal. Upon identifying a far end audio (“FEA”) delay based on the first calibration audio signal and the second calibration audio signal, a first mixed audio data containing the first shared audio data and first FEA data is fetched from an audio buffer. The first FEA data is subsequently removed or extracted from the mixed audio data in response to the FEA delay.

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: An adaptive filter includes a frequency domain adaptation block that analyzes a statistic of coefficient movement in the frequency domain. The adaption block adjusts, in the frequency domain, a parameter (step size or leakage factor) that affects speed of convergence of the adaptive filter based on the analyzed statistic of filter coefficient movement. The filter includes an associated coefficient, statistic of coefficient movement, and parameter for each frequency bin. The coefficients may be complex numbers, and separate real and imaginary statistics and parameters are maintained. The statistic may be direction counts of the filter coefficient movement. The step size may be adjusted to a predetermined minimum value when the current direction of movement of the filter coefficient is different than the predominant direction and otherwise the step size is adjusted approximately proportionally to an amount of predominance by a value based on a direction count of the filter coefficient movement.

Abstract: A method of suppressing residual echo includes calculating a spectral mismatch of an acoustic echo canceler based upon a program content signal and a residual signal; determining a set of filter coefficients based at least in part upon a selected spectral mismatch; filtering the residual signal based upon the set of filter coefficients; freezing the calculation of the selected spectral mismatch in response to detecting a double talk condition in the residual signal; waiting a predetermined hold period in response to detecting that the double talk condition has ended; and, after the predetermined hold period, resuming the calculation of the spectral mismatch based upon the program content signal and the residual signal.

Abstract: An echo cancellation method includes steps of (a) extracting phase-distortion estimates, (b) reconstructing an echo signal, (c) generating a clean signal, and (d) producing a primary signal. Step (a) includes extracting, from a first phase signal, a plurality of phase-distortion estimates, the first phase signal having been estimated from an echo-corrupted signal received at a first coherent transceiver of a coherent optical network. Step (b) includes reconstructing an echo signal from the plurality of phase-distortion estimates and a transmitted signal transmitted by the first coherent transceiver. Step (c) includes generating a clean signal as a difference between the reconstructed echo signal and the first phase signal. Step (d) includes producing a primary signal by mapping each of a plurality of clean-phase estimates of the clean signal to one of a plurality of constellation symbols associated with a modulation scheme of the primary signal.

Abstract: A method for controlling volume in an apparatus having a speaker and a microphone includes receiving, at the microphone, external noise and speech of a user, and calculating sound pressure of the noise received by the microphone. The method further includes performing exception processing of the sound pressure of some or all of the noise using the calculated sound pressure and one of a speech utterance state, a speech receiving state, or a temporal length state, of the noise, mapping volume of the speech in response to the sound pressure of the external noise, synthesizing speech guidance into a sound file, outputting the sound file, via the speaker, according to the mapped volume.

Abstract: A system configured to improve echo cancellation for nonlinear systems. The system generate reference audio data by isolating portions of microphone audio data that correspond to playback audio data. For example, the system may determine a correlation between the playback audio data and the microphone audio data in individual time-frequency bands in a frequency domain. In some examples, the system may substitute microphone audio data associated with output audio for the playback audio data. The system may generate the reference audio data based on portions of the microphone audio data that have a strong correlation with the playback audio data. The system may generate the reference audio data by selecting these portions of the microphone audio data or by performing beamforming. This results in precise time alignment between the reference audio data and the microphone audio data, improving performance of the echo cancellation.

Abstract: A system configured to improve audio processing by adaptively selecting target signals based on current system conditions. For example, a device may select a target signal based on a highest signal quality metric when only the local speech is present (e.g., during near-end single-talk conditions), as this maximizes an amount of energy included in the output audio signal. In contrast, the device may select the target signal based on a lowest signal quality metric when only the remote speech is present (e.g., during far-end single-talk conditions), as this minimizes an amount of energy included in the output audio signal. In addition, the device may track positions of the local speech and the remote speech over time, enabling the device to accurately select the target signal when both local speech and remote speech is present (e.g., during double-talk conditions).

Abstract: A method for echo cancellation based on microphone signal correlation is disclosed.

Abstract: Techniques for acoustic echo cancellation are described herein. In an example embodiment, a system comprises a speaker, a microphone array with multiple microphones, a beamformer (BF) logic and an acoustic echo canceller (AEC) logic. The speaker is configured to receive a reference signal. The BF logic is configured to receive audio signals from the multiple microphones and to generate a beamformed signal. The AEC logic is configured to receive the beamformed signal and the reference signal. The AEC logic is also configured to compute a vector of bias coefficients multiple times per time frame, to compute a background filter coefficient based on the vector of bias coefficients, to apply a background filter to the reference signal and the beamformed signal based on the background filter coefficient, to generate a background cancellation signal, and to generate an output signal based at least on the background cancellation signal.

Abstract: A circuit for mitigating electric shock including an external impedance detection circuit and a test and holdoff circuit. The external impedance detection circuit detects a presence of an external impedance, such as by detecting a relative change in voltage from a startup condition and a test condition. The test and holdoff circuit inhibits operation of a power converter which delivers power to be consumed by a load. The startup condition is defined by mains power applied to the circuit with negligible power consumed by a load. The test condition is defined by non-zero power delivered to the load. According to another aspect, the external impedance detection circuit measures an input voltage using a high input power which is greater than a shock hazard threshold at a duration less than a threshold time duration and determines the presence of the external impedance based on low pass filters having different time constants.

Abstract: A system identification device for performing fast real-time identification for a system from input/output data includes a filter robust to disturbance, by setting the maximum energy gain from the disturbance to a filter error, as an evaluation criterion, smaller than a given upper limit. The filter estimates a state estimation value of a state of the system.

Abstract: A method for substantially eliminating the effect of mechanical vibration on an audio input to a speakerphone system is provided herein, the method comprising: receiving an input sound acoustic signal at a microphone (mic); converting the received input sound acoustic signal into an input sound electrical signal, and outputting the same as a mic output signal; receiving mechanical vibrations at a mechanical vibration sensor (MVS); converting the received mechanical vibrations into a mechanical vibration error signal, and outputting the same as an MVS output signal; and generating a speakerphone system output signal by subtracting the MVS output signal from the mic output signal.

Abstract: An acoustic echo path change detector provides a monitoring process in an acoustic echo canceler that removes echo from a microphone signal using an adaptive echo path model that generates an echo estimate from a playback signal. The acoustic echo canceler removes the echo estimate from the microphone signal to provide an echo-canceled output signal. The path change detector receives the microphone signal, the echo estimate and the output signal and determines a rate of change of one or more statistical values dependent on the microphone signal, the echo estimate and the output signal. If the rate of change exceeds a threshold value, the echo path change detector generates an indication that causes a supervisory process to change adaptation of the adaptive echo path model to increase responsiveness to the change in the acoustic echo path, e.g., by increasing the step size.

Abstract: The systems, devices, and processes described herein may identify a beam of a voice-controlled device that is directed toward a reflective surface, such as a wall. The beams may be created by a beamformer. An acoustic echo canceller (AEC) may create filter coefficients for a reference sound. The filter coefficients may be analyzed to identify beams that include multiple peaks. The multiple peaks may indicate presence of one or more reflective surfaces. Using the amplitude and the time delay between the peaks, the device may determine that it is close to a reflective surface in a direction of the beam.

Abstract: According to one embodiment, a spatial position measurement apparatus, includes: a transmission unit configured to transmit an ultrasonic wave accompanying with a transmission source identifiable from three or more transmission sensors provided on a first object; a detection unit configured to detect the ultrasonic wave received by two or more reception sensors provided on a second object; a distance calculation unit configured to calculate distances between the transmission sensors and the reception sensors based on propagation time of the ultrasonic wave; and a coordinate calculation unit configured to calculate, in a coordinate system where a position of one group out of a group of the transmission sensors and a group of the reception sensors is fixed, positional coordinates of another group based on the distances.

Abstract: Techniques for improving acoustic echo cancellation to attenuate an echo signal generated by a loudspeaker included in a device are described. A system may determine a loudspeaker canceling beam (LCB) (e.g., fixed beam directed to the loudspeaker) and may use the LCB to generate LCB audio data that corresponds to the echo signal. For example, based on a configuration of the loudspeaker relative to microphone(s) of the device, the system may perform simulation(s) to generate a plurality of filter coefficient values corresponding to the loudspeaker. By subtracting the LCB audio data during acoustic echo cancellation, the system may attenuate the echo signal even when there is distortion or nonlinearity or the like caused by the loudspeaker. In some examples, the system may perform acoustic echo cancellation using the LCB audio data and playback audio data.

Abstract: An aspect of this disclosure relates to noise and/or echo suppression for a device in which noise and echo suppression are adaptively determined as noise and echo change in an environment that surrounds the device. An aspect can use a skewed maximal ratio combining technique or a spatial filter with coefficients that are adaptively determined based on a perceptually selected target ratio that is compared to a ratio of sound energies/levels based on a pair of the coefficients. Another aspect relates to the use of information in one frequency band to perform additional noise and/or echo suppression in one or more adjacent frequency bands.

Abstract: A method for canceling acoustic far-end audio echo content includes high-pass filtering audio content received from a far end and playing the high-pass filtered audio content through a speaker, examining audio content captured by a microphone to detect the presence of audio content in a low-frequency sub band after subtracting high-pass filtered audio content from the audio content captured by the microphone using a least mean square (LMS) adaptive filter. If audio content in a low-frequency sub band is detected in the audio content captured by the microphone, freezing adaptation of the LMS filter and sending to the far end the audio content captured by the microphone after subtracting, and if audio content in a low-frequency sub band is not detected in the audio content captured by the microphone, enabling adaptation of the LMS filter and sending to the far end the audio content captured by the microphone after subtracting.

Abstract: Techniques related to active acoustic echo cancellation are discussed. Such techniques may include generating an audio output signal having a portion thereof corresponding to a first audio frequency range to negate a response of an audio input device to an output from a speaker in a second audio frequency range at a response negation rate and decimating an audio input signal based on the response negation rate to generate a resultant audio input signal.

Abstract: Techniques for real-time audio communication including supplying an initial audio signal to an audio signal processor configured to process the initial audio signal and output a processed audio signal to an audio output means, obtaining a loopback audio signal corresponding to the processed audio signal, generating a plurality of audio features for the initial audio signal and the loopback audio signal, generating a similarity measure estimating a similarity of the initial audio signal to the first loopback audio signal based on at least the plurality of audio features, adjusting an adaptation rate for an audio signal processing operation based on at least the similarity measure, and controlling, based on at least the adjusted adaptation rate, an echo cancellation process for cancelling an estimated echo from a near-end audio signal received via an audio input means in proximity to the audio output means.

Abstract: Communication apparatus and devices for surgical robotic systems are described. The communication apparatus can include a user console in communication with a communication device having a surgical tool. The communication device can include a microphone to convert a sound input into an acoustic input signal. The communication device can transmit the acoustic input signal to the user console for reproduction as a sound output for a remote operator. The surgical tool can include an endoscope having several microphones mounted on a housing. The surgical tool can be a sterile barrier having a microphone and a drape. The microphone(s) of the surgical tools can face a surrounding environment such that a tableside staff is a source of the sound input that causes the sound output, and a surgeon and the tableside staff can communicate in a noisy environment. Other embodiments are also described and claimed.

Abstract: A method of enhancing an audio signal, the method comprising: receiving a plurality of input audio signals from a plurality of microphones; for each of the plurality of input audio signals, generating at an echo cancellation module, at least one output signal, the at least one output signal comprising one or more of an echo cancelled signal, a post-filter signal and a filter tap signal; analysing the plurality of input audio signals and/or the respective at least one output signal to determine a condition at each of the plurality of microphones; selecting one of the at least one output signals based on the determined condition at each of the plurality of microphones; and generating an echo suppressed audio signal by suppressing echo in an audio signal derived from one or more of the plurality of microphones using the selected one of the at least one output signal.