Abstract: Circuitry for noise cancellation in a headset worn by a user, the circuitry comprising: an input for receiving one or more heartbeat signals representative of a heartbeat of the user; a processor for generating a heartbeat noise cancellation signal based on the one or more heartbeat signal, the heartbeat noise cancellation signal to cancel a noise associated with the heartbeat in the user's ear; an output for outputting the heartbeat noise cancellation signal to a transducer of the headset.

Abstract: Method for setting parameters for individual adaptation of an audio signal, including: performing a first listening test with the substeps: playing a plurality of first audio signals having different levels; obtaining feedback per frequency range from an individual which of the plurality of first acoustic signals is above an individual listening threshold; and using the lowest level of the different levels for which feedback is available as a level for the individual listening threshold per frequency range; performing adaptation of a second audio signal with the substeps: playing the second audio signal according to a total volume level considering a sound adaptation characteristic map; and varying the sound adaptation characteristic wherein the levels for the individual listening thresholds are used as minimum output levels in the sound adaptation characteristic map.

Abstract: Aspects of the invention are directed towards a system and method for noise suppression. One or more embodiments of the invention describe the method comprising steps of receiving noise by a microphone and deriving a noise signal from the noise, the noise produced by a fan of a fire/smoke detection unit while drawing air and detecting the speed of the fan of the fire/smoke detection unit. The method further describes steps of amplifying the noise signal by the amplifier received from the microphone and producing an anti-phase noise signal by shifting a phase of the amplified noise signal received from the amplifier wherein the anti-phase noise signal is dependent on the detected speed of the fan. The method further describes steps of outputting the anti-phase noise signal through a speaker to suppress the noise produced by the fan of the fire/smoke detection unit while drawing the air.

Abstract: A method, a computer program product, and a computer system predictively compensate for expected audio communication issues. The method includes determining conditions of a first device associated with a user. The method includes determining, based on the conditions, a first signal strength to a first network that the first device is currently connected in which to perform a communication and a second signal strength to a second network that the first device is configured to utilize in performing the communication. As a result of each of the first signal strength and the second signal strength not satisfying a minimum threshold individually, the method includes generating an overall signal having an overall signal strength by layering the first network over the second network. The overall signal strength has a comparatively greater signal strength than the first signal strength and the second signal strength. The method includes performing the communication using the overall signal.

Abstract: An apparatus and method of transmission control for an audio device. The audio device uses sources other than the microphone to determine nuisance, and uses this to calculate a gain as well as to make the transmit decision. Using the gain results in a more nuanced nuisance mitigation than using the transmit decision on its own.

Abstract: A system may include an electromagnetic load capable of generating a haptic event and a haptic processor configured to receive at least one first parameter indicative of a desired perception of the haptic event to a user of a device comprising the electromagnetic load, receive at least one second parameter indicative of one or more characteristics of the device, and process the at least one first parameter and the at least one second parameter to generate a driving signal to the electromagnetic load in order to produce the desired perception to the user despite variances in the device that cause an actual perception of the haptic event to vary from the desired perception.

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

Abstract: An image capture device may include a microphone, a vibration sensor, and a processor. The microphone may obtain a microphone signal that includes an acoustic signal portion and a mechanical noise portion. The vibration sensor may obtain a vibration signal. The processor may upsample the vibration signal. The processor may determine a correlation value. The correlation value may be based on the microphone signal, the upsampled vibration signal, or both. The processor may determine filter coefficients. The filter coefficients may be determined on a condition that the correlation value is above a threshold. The filter coefficient may be based on the upsampled vibration signal. The processor may filter the vibration signal based on the filter coefficients to remove the mechanical noise portion and obtain a processed microphone signal. The processor may output the processed microphone signal.

Abstract: Adaptive noise cancellation systems and methods comprise a reference sensor operable to sense environmental noise and generate a corresponding reference signal, an error sensor operable to sense noise in a noise cancellation zone and generate a corresponding error signal, a noise cancellation filter operable to receive the reference signal and generate an anti-noise signal to cancel the environmental noise in the cancellation zone, an adaptation module operable to receive the reference signal and the error signal and adaptively adjust the anti-noise signal. The adaptation module includes a noise amplification control module operable to adaptively control noise amplification in at least one hiss region of the anti-noise signal, while achieving cancellation in non-hiss regions of the anti-noise signal.

Abstract: A system for generating adaptive notifications including processors and memories storing instructions that configure the processors to perform operations. The operations may include generating an item list for notifications to a client ID, selecting a first item from the item list for notification, the first item being selected having a highest priority score and a lowest lifecycle timer, encoding a first landing page by encoding a product code and a tracking cookie in a URL configured for redirection to a first webpage by a redirect server, and transmitting a first notification with a payload including the first landing page and a first message. The operations may also include after transmitting the first notification, modifying priority scores and updating lifecycle timers based on determining whether the client device visited the first landing page. Further, the operations may also include selecting a second item from the item list for notification.

Abstract: In one example, a headset obtains, from a first microphone on the headset, a first audio signal including a user audio signal and an anisotropic background audio signal. The headset obtains, from a second microphone on the headset, a second audio signal including the user audio signal and the anisotropic background audio signal. The headset extracts, from the first audio signal and the second audio signal, using a first adaptive filter, a reference audio signal including the anisotropic background audio signal. Based on the reference signal, the headset cancels, using a second adaptive filter, the anisotropic background audio signal from a third audio signal derived from the first and second audio signals to produce an output audio signal. The headset provides the output audio signal to a receiver device.

Abstract: This application describes a noise reduction apparatus (800) for a microphone device (100) having an acoustic port (110). The apparatus has a spectrum peak detect block (301) for receiving a microphone signal (SMIC) and determining, from the microphone signal, at least one characteristic of a resonance peak (202) associated with the acoustic port of the microphone. The at least one characteristic may comprise a resonance frequency (fH) and/or quality factor (QH). A noise reduction block (801) is configured to process the microphone signal based on the resonance characteristic so as to reduce noise in the processed microphone signal due to said resonance. The noise reduction block may apply a function which is the inverse of the determined resonance characteristic.

Abstract: An acoustic interference cancellation system that combines acoustic echo cancellation and an adaptive beamformer to cancel acoustic interference from an audio output. The system uses a fixed beamformer to generate a target signal in a look direction and an adaptive beamformer to generate noise reference signals corresponding to non-look directions. The noise reference signals are used to estimate acoustic noise using an acoustic interference canceller (AIC), while reference signals associated with loudspeakers are used to estimate an acoustic echo using a multi-channel acoustic echo canceller (MC-AEC). The system cancels the acoustic echo and the acoustic noise simultaneously by adding the estimate of the acoustic noise and the estimate of the acoustic echo to generate an interference reference signal and cancelling the interference reference signal from the target signal. The system jointly updates adaptive filters for the AIC and the MC-AEC logic to improve a robustness of the system.

Abstract: A loudspeaker drive circuit uses a dynamic range compressor to implement a non-linear gain function between the input signal to the dynamic range compressor and an output signal from the dynamic range compressor. The output is used to drive a loudspeaker, and the operating parameters of the dynamic range compressor are varied in dependence on a criterion, such as the estimated voice coil temperature, the power consumption or the acoustical distortion.

Abstract: Method for improving noise suppression for ASR starts with a microphone receiving an audio signal including speech signal and noise signal. In each frame for frequency band of audio signal, a noise estimator detects ambient noise level and generates noise estimate value based on estimated ambient noise level, variable noise suppression target controller generates suppression target value using noise estimate value and logistic function, a gain value calculator generates a gain value based on suppression target value and noise estimate value, and combiner enhances the audio signal by the gain value to generate a clean audio signal in each frame for all frequency bands. Logistic function models desired noise suppression level that varies based on ambient noise level. Variable level of noise suppression includes low attenuation for low noise levels and progressively higher attenuation for higher noise level. Other embodiments are also described.

Abstract: A microphone array includes an n microphone units (where n is an integer equal to or larger than three). A first one of the microphone units includes three microphones ((14-1 through 14-3)) arranged in line at equal intervals. An m-th microphone unit (where m is a positive integer expressed by 1<m<n) includes the microphones at either end of an (m?1)-th microphone unit, and another microphone spaced from the microphone at one end of the (m?1)-th microphone unit by a distance substantially equal to the distance between the microphones at either end of the (m?1)-th microphone unit, and disposed at a location on the side of the (m?1)-th microphone unit opposite to the side where the microphone at the one end at the one end of the (m?1)-th microphone unit is disposed.

Abstract: Embodiments of the present invention pertain to reducing or eliminating speech recognition error when background noise is detected at a caller's location. For example, when background noise is detected at the caller's location, the caller may be prompted to use dual-tone multi-frequency (DTMF).

Abstract: Speech recognition error may be reduced or eliminated when background noise is detected at a customer's location. For example, when background noise is detected at the customer's location, the customer may be prompted to use dual-tone multi-frequency (DTMF).

Abstract: A hearing assistance device includes two transducers which react to a characteristic of an acoustic wave to capture data representative of the characteristic. The device is arranged so that each transducers is located adjacent a respective ear of a person wearing the device. A signal processor processes the data to provide relatively more emphasis of data representing a first sound source the person is facing over data representing a second sound source the person is not facing. At least one speaker utilizes the data to reproduce sounds to the person. An active noise reduction system provides a signal to the speaker for reducing an amount of ambient acoustic noise in the vicinity of the person that is heard by the person.

Abstract: Methods, systems, and apparatus, including computer programs encoded on a computer storage medium, for receiving an input from a call center agent indicating a symptom of a quality of an electronic communication between the agent and a caller. Identifying a cause of the quality of the electronic communication based on the input. And, sending instructions to adjust the cause of the quality of the electronic communication.

Abstract: Method of dynamically adapting playback volume on electronic device starts with processor receiving first user input and first portion of audio content. First user input signals to device to increase or decrease volume of sound output. Processor determines first loudness metric corresponding to first portion of audio content when first user input is received. First loudness metric is measure of loudness of first portion of audio content being outputted. Processor then stores in memory first loudness metric in association with first user input. Memory stores history of loudness metrics in association with user inputs. Processor then determines second loudness metric that is measure of loudness of second portion of audio content that is received and determines second user input associated with second loudness metric using history. Processor generates control signal to automatically control volume of sound output by device corresponding to second user input. Other embodiments are also described.

Abstract: Embodiments of the present invention pertain to reducing or eliminating speech recognition error when background noise is detected at a caller's location. For example, when background noise is detected at the caller's location, the caller may be prompted to use dual-tone multi-frequency (DTMF).

Abstract: Speech recognition error may be reduced or eliminated when background noise is detected at a customer's location. For example, when background noise is detected at the customer's location, the customer may be prompted to use dual-tone multi-frequency (DTMF).

Abstract: Communications equipment for use by a call centre to enable communications between the call centre and one or more user devices is provided, with the communications equipment comprising audio processing circuitry for processing audio signals received from the user devices over a communications network, wherein the processing circuitry is configured to perform acoustic echo cancellation and/or noise suppression and/or dereverberation processing on the received audio signals to cancel acoustic echoes and/or suppress noise and/or suppress reverberation that is present in the audio signals received from the user device.

Abstract: Systems and methods for removing noise from an audible signal are provided. More particularly, a vibration sensor is used to obtain a vibration signal from an environment including a communication device. The signal from the vibration sensor is combined with a signal from a microphone associated with the communication device, to create a modified audible signal. More particularly, a filtering or subtraction process can be performed with respect to the audible signal, at a time corresponding to an event detected as part of the vibration signal. The resulting modified audible signal can have reduced noise as compared to the original audible signal.

Abstract: A system, method, and computer program product are provided for cleaning an audio segment. For a given audio segment, an offset amount is calculated where the audio segment is maximally correlated to the audio segment as offset by the offset amount. The audio segment and the audio segment as offset by the offset amount are averaged to produce a cleaned audio segment, which has had noise features reduced while having signal features (such as voiced audio) enhanced.

Abstract: An apparatus is provided that includes first and second ICs configured to communicate using a plurality of differential signal lines. The apparatus includes a common mode suppression circuit having a plurality of common mode voltage adjustment circuits, each configured to provide a low impedance path for common mode signals and a high impedance path for differential AC signaling, thereby suppressing the effect of common mode transients between the voltage domains. The plurality of common mode voltage adjustment circuits each have components that are impedance matched up to an impedance-tolerance specification. The common mode suppression circuit also includes an AC coupling circuit configured to be less dependent on impedance mismatch, beyond the impedance-tolerance specification, by cross coupling the impedance differentials from each of the differential signal lines through the AC coupling circuit and to one of the common mode voltage adjustment circuits.

Abstract: Techniques are described herein that suppress noise using multiple sensors (e.g., microphones) of a communication device. Noise modeling (e.g., estimation of noise basis vectors and noise weighting vectors) is performed with respect to a noise signal during operation of a communication device to provide a noise model. The noise model includes noise basis vectors and noise coefficients that represent noise provided by audio sources other than a user of the communication device. Speech modeling (e.g., estimation of speech basis vectors and speech weighting) is performed to provide a speech model. The speech model includes speech basis vectors and speech coefficients that represent speech of the user. A noisy speech signal is processed using the noise basis vectors, the noise coefficients, the speech basis vectors, and the speech coefficients to provide a clean speech signal.

Abstract: A Pre-distortion mechanism for transmit path non-linearity in xDSL AFE is disclosed in the present invention. The AFE includes a line driver and a pre-distortion signal generator. The line driver receives an input differential signal and generates an output differential signal. The input differential signal includes a first input signal and a second input signal. The output differential signal includes a first output signal and a second output signal. The line driver receives the first input signal to generate the first output signal and receives the second input signal to generate the second output signal. The pre-distortion signal generator is coupled to input ends and output ends of the line driver.

Abstract: A call-center has agents using headsets, which are connected to a private business exchange (PBX). Noise cancellation occurs in a noise filter, within or connected to the PBX. Noise cancellation for a particular agent's conversation is achieved by receiving the voice signals from neighboring agents' headsets and by using adaptive noise cancellation in the noise filter to remove the other agents' conversations from the particular agent's conversation. Microphones may also be placed at other noise sources, such as HVAC equipment, so that offending noises are accurately received at the noise filter and removed from the agents' conversations.

Abstract: Systems and methods are provided for a media device that controls input and output characteristics based on one or more associated conditions.

Abstract: Noise reduction is provided to audio captured by a headset by employing spatially separated microphones provided by a headset and a mobile phone. Primary audio is captured by a headset microphone and includes both voice audio and ambient noise. Secondary audio is captured by a mobile phone microphone and includes ambient audio. Noise reduction is performed using the primary and secondary audio to generate a noise-reduce audio.

Abstract: A method and computing system for suppressing noise in an audio signal, comprising: receiving the audio signal at signal processing means; determining that another signal is input to the signal processing means, the input signal resulting from an activity which generates noise in the audio signal; and selectively suppressing noise in the audio signal in dependence on the determination that the input signal is input to the signal processing means to thereby suppress the generated noise in the audio signal.

Abstract: The present invention relates to a method as well as to a computing device (20) for editing a noise-database (13) containing noise information, said noise information being derived from noise signals within an audio stream (19). In order to enhance possibilities to create and utilize context information which emerge from tracking noise signals from an audio stream, for example a telephone call, the above method is characterized by the following steps: A) in a localizing step (14), determining geographical data of the location the noise signals origin from; B) in an analyzing step (15), analyzing the noise signals with reference to the noise content; C) in a linking step, linking the analyzed noise signals to said geographical data to create noise information; D) in a storing step, storing said noise information within said noise-database (13).

Abstract: This document discusses, among other things, systems and methods for active noise cancellation. One example system includes a digital ANC circuit configured to receive first audio information from a first microphone and to produce an a digital anti-noise signal configured to attenuate noise sensed by the first microphone; an analog ANC circuit configured to receive second audio information from a second microphone and to produce an analog anti-noise signal configured to attenuate noise sensed by the second microphone; and wherein the system is configured to receive an intended audio signal and to provide an output signal for a speaker using the intended audio signal, the analog anti-noise signal, and the digital anti-noise signal.

Abstract: Technologies are generally described for a system for measuring a quality of experience (QoE). In some examples, a quality of experience (QoE) measuring device may include a background noise detecting unit configured to measure a background noise around a near-end talker, and a decision unit configured to determine whether a double talk event detected by a double talk detector is caused by the background noise around the near-end talker.

Abstract: Systems and methods for removing noise from an audible signal are provided. More particularly, a vibration sensor is used to obtain a vibration signal from an environment including a communication device. The signal from the vibration sensor is combined with a signal from a microphone associated with the communication device, to create a modified audible signal. More particularly, a filtering or subtraction process can be performed with respect to the audible signal, at a time corresponding to an event detected as part of the vibration signal. The resulting modified audible signal can have reduced noise as compared to the original audible signal.

Abstract: A correction spectrum calculation unit 6 obtains a correction spectrum by smoothing an estimated noise spectrum in accordance with the degree of its variations, and a suppression quantity limiting coefficient calculation unit 7 decides a suppression quantity limiting coefficient from the correction spectrum. A suppression quantity calculation unit 9 obtains a suppression coefficient based on the suppression quantity limiting coefficient, and the spectrum suppression unit 10 carries out amplitude suppression of spectral components of an input signal.

Abstract: A method and apparatus for finding an estimate of the delay of a signal travelling between two points. A quantity is evaluated from the signal at a final number of time instants, at both a reference point and a reception point. The values are quantized by comparison with a threshold adapted to a typical magnitude of the quantity. If the quantized values from the reception point are shifted back by the true delay with respect to the quantized values from the reference point, then certain co-occurrences of quantized values have very low probability. Hence, the best delay estimate is that shift which yields the least number of low-probability co-occurrences.

Abstract: A method and an electronic device for communication quality improvement based on ambient noise sensing are provided for improving communication quality. The method includes: during a call, dynamically setting a noise reduction mode in response to the varying of an ambient noise amount of at least one audio signal received by the electronic device. The setting step includes: during the call, dynamically determining whether the ambient noise amount indicates the needs of the application of noise reduction. If the ambient noise amount indicates the needs of the application of noise reduction, then one of a plurality of noise reduction levels is selected according to the ambient noise amount, and the noise reduction mode is set according to the selected noise reduction level, wherein the noise reduction levels include at least two levels corresponding to two different amounts of noise reduction.

Abstract: A noise suppression system is applicable to a first phone, to reduce noise received by a microphone of the first phone. The microphone receives sound signals with different frequencies and converts the sound signals to electrical signals. The noise suppression system includes a selection circuit, an amplifying circuit, and a processing circuit. The selection circuit selects special electrical signals with frequency between 65 hertz to 1.1 kilohertz from the electrical signals. The amplifying circuit amplifies the selected electrical signals. The processing circuit stores a first preset value and a second preset value less than the first preset value. If the value of the amplified electrical signal is greater than or equal to the first preset value, or less than or equal to the second preset value, the processing circuit outputs the amplified electrical signal.

Abstract: A method for maintaining transmit audio quality under harsh environmental conditions, the method includes receiving audio signals into a microphone of a portable communication device and determining at least one parameter associated with the received audio signals. In accordance with an embodiment, the at least one parameter is compared with a received audio parameter threshold. When the at least one parameter falls outside of the received audio parameter threshold, the audio routing is switched from the microphone to the loudspeaker. Subsequent communication may revert back to the microphone or remain at the loudspeaker depending on monitored audio conditions.

Abstract: An adaptive hybrid system is coupled to a loop for adjusting trans-hybrid loss. The system comprises a fixed portion comprising a first receiver transfer function block and a first hybrid transfer function block. The fixed portion is configured to receive a far-end signal and mitigate frequency dependent attenuation experienced by the far-end signal. The system also comprises a variable portion comprising a second receiver transfer function block and a second hybrid transfer function block configured to subtract a transmit echo from the received far-end signal.

Abstract: A method for removing a noise signal from an input signal that is received. In the method, the amount of energy of the input signal is detected, a noise signal included in the input signal is estimated, an intermediate output signal is generated by removing the estimated noise signal from the input signal, and a final output signal is generated by amplifying the amount of energy of the intermediate output signal based on a difference between the amount of energy of the intermediate output signal and the amount of energy of the input signal.

Abstract: In one aspect thereof the invention provides a method for noise suppression of a speech signal that includes, for a speech signal having a frequency domain representation dividable into a plurality of frequency bins, determining a value of a scaling gain for at least some of said frequency bins and calculating smoothed scaling gain values. Calculating smoothed scaling gain values includes, for the at least some of the frequency bins, combining a currently determined value of the scaling gain and a previously determined value of the smoothed scaling gain. In another aspect a method partitions the plurality of frequency bins into a first set of contiguous frequency bins and a second set of contiguous frequency bins having a boundary frequency there between, where the boundary frequency differentiates between noise suppression techniques, and changes a value of the boundary frequency as a function of the spectral content of the speech signal.

Abstract: The sidetone noise level in a wireless or wired telephone, for example, is reduced by adjusting both noise and signal levels and/or by improving the signal-to-noise ratio of such signals.

Abstract: A circuit apparatus and a method for detecting an earphone in a portable terminal are provided. The apparatus includes an earjack, a first switch, a second switch, a comparator, and a main chip. The earjack outputs a signal informing whether an earphone is inserted, provides a signal input from a third pole of the inserted earphone to the first switch, and provides a signal input from a fourth pole to the second switch. The first switch outputs a signal provided from the earjack to one of a comparator and a ground depending on an input control signal. The second switch outputs a signal provided from the earjack to one of the comparator and the ground depending on the input control signal. The comparator compares a voltage of a signal provided from one of the first switch and the second switch with a reference voltage to output a signal representing a comparison result.

Abstract: A sound processor includes a conversion unit converts a reference sound signal corresponding to a base of sound to be output and an observation sound signal based on each of sound signals output by a plurality of sound receiving units into frequency components, an echo suppression unit estimates echo derived from sound based on a converted reference sound signal and suppressing the estimated echo in a converted observation sound signal, a noise suppression unit estimates noise based on an arrival direction of sound and suppressing the estimated noise in the converted observation sound signal and an integrating process unit suppresses, with respect to each frequency component, echo and noise in the converted sound signal based on a observation sound signal obtained after echo suppression and a observation sound signal obtained after noise suppression.

Abstract: A power supply circuit including a voltage regulator and a headphone driving circuit employing the same are disclosed. In accordance with the present invention, the power supply circuit may provide stable power supply to the amplifier since the power supply circuit generates the constant voltage +Vreg and the constant voltage ?Vreg from +VDD robust to stronger than a power noise, and the balanced signal path for isolating the noise of the input signal to be amplified is not required, thereby simplifying the constitution of the circuit.

Abstract: In accordance with a non-limiting example, an electrical protection device includes a fuse housing and a plurality of fuses carried by the fuse housing and arranged in a plurality of Tip/Ring fuse pairs. The Tip/Ring fuse pairs are arranged in vertical and horizontal orientation of Tip/Ring fuse pairs and spaced to each other such that differential crosstalk among the Tip/Ring fuse pairs is cancelled.