Abstract: Techniques are described for reducing sensitivity to non-acoustic stimuli. In some embodiments, differential beamforming is applied to microphone signals generated based on responses of microphones to an acoustic stimulus and a non-acoustic stimulus. Compensated signals can be generated based on the microphone signals such that the compensated signals are in phase with respect to the acoustic stimulus. The non-acoustic stimulus is detectable by comparing a first signal to a second signal to determine that one signal has a greater instantaneous magnitude. The first signal can be a beamformed signal or signal derived therefrom, and the second signal can be an average of the compensated signals or signal derived therefrom. An output audio signal can be generated by switching or cross fading between the beamformed signal and a noise-reduced signal such that a contribution of the noise-reduced signal is increased and a contribution of the beamformed signal is decreased.

Abstract: An amplifier unit for operating a piezoelectric loudspeaker or microphone includes an audio amplifier and a detection unit, which is configured to detect whether a sound transducer coupled to the amplifier unit is a piezoelectric sound transducer or a dynamic sound transducer. The amplifier unit is configured in such a way that, after a sound transducer has been coupled, the amplifier unit sends a test signal to the coupled sound transducer. Moreover, a sound-generation unit includes a sound transducer and an amplifier unit, which amplifies an audio signal and feeds it to the sound transducer.

Abstract: An outcome-based receiver beam tuning is provided. A base station device can transmit a reference signal with a group of symbols corresponding to different transmit beams, while the reference signal instructs the user equipment to not perform receive beam sweeping. The user equipment device can report the transmit beam with the highest signal strength. The base station device can then transmit a reference signal using the transmit beam with the highest signal strength, while also informing the user equipment device to perform receive beam sweeping. By comparing the receive beam with the highest signal strength to the signal strength of the best transmit beam, the base station device can determine the density (e.g. periodicity) of CSI-RS with repetition “ON” transmission.

Abstract: This disclosure describes, in part, techniques for detecting that a microphone of a voice-enabled device is unusable due to noise levels around the microphone, and indicating to a user that the device has become unusable due to, for example, buffeting. The device may analyze audio signals to determine that the noise levels are too high to perform speech-processing techniques to detect voice. The device may utilize speakers to output sounds, and/or LEDs to output light, to indicate noise levels and/or that the device is unable to detect voice due to buffeting. In this way, the device may indicate that a user needs to speak loudly to be heard, move their device, and/or reduce sound in the environment to use the speech-processing abilities of the device.

Abstract: Provided are methods and systems for noise suppression within multiple time-frequency points of spectral representations. A multi-feature cluster tracker is used to track signal and noise sources and to predict signal versus noise dominance at each time-frequency point. Multiple features, such as binaural and monaural features, may be used for these purposes. A Gaussian mixture model (GMM) is developed and, in some embodiments, dynamically updated for distinguishing signal from noise and performing mask-based noise reduction. Each frequency band may use a different GMM or share a GMM with other frequency bands. A GMM may be combined from two models, with one trained to model time-frequency points in which the target dominates and another trained to model time-frequency points in which the noise dominates. Dynamic updates of a GMM may be performed using an expectation-maximization algorithm in an unsupervised fashion.

Abstract: A condenser microphone includes a condenser microphone unit and a piezoelectric element. The piezoelectric element is disposed so as to generate piezoelectric signals in response to vibration causing the unit to generate vibratory noise signals. The piezoelectric signals are inputted through a low-pass filter and a level adjuster circuit to the unit to drive a diaphragm of the unit. The vibratory noise signals generated by the vibration in the unit are canceled with the piezoelectric signals generated by the piezoelectric element.

Abstract: In an embodiment of the invention, the voice coil of an electro dynamic transducer is protected against thermal overload by estimating the temperature of a magnet in the electro dynamic transducer. When a power limit based on the temperature of the magnet and on a predetermined voice coil temperature limit is reached by an audio signal, the power applied to the voice coil is reduced.

Abstract: A signal processing apparatus includes: one or more detection means for detecting movement of a diaphragm of a speaker in correspondence with feedback methods that are different feedback methods; analog-to-digital conversion means for converting one or more detection signals acquired by the detection means into a digital form; feedback signal generating means for generating feedback signals corresponding to the feedback methods using the digital detection signals; synthesis means for combining an audio signal to be output as a driving signal of the speaker with the feedback signals; correction equalizer means for setting an equalizing characteristic to allow a sound reproduced by the speaker to have a target frequency characteristic by changing the digital audio signal; feedback operation setting means for setting feedback methods in which a feedback operation up to combining the audio signal with the feedback signal is performed and the feedback operation is not performed equalizing characteristic changing a

Abstract: A system for compensating and driving a loudspeaker includes an open loop loudspeaker controller that receives and processes an audio input signal and provides an audio output signal. A dynamic model of the loudspeaker receives the audio output signal, and models the behavior of the loudspeaker and provides predictive loudspeaker behavior data indicative thereof. The open loop loudspeaker controller receives the predictive loudspeaker behavior data and the audio input signal, and provides the audio output signal as a function of the audio input signal and the predictive loudspeaker behavior data.

Abstract: A system for compensating and driving a loudspeaker includes an open loop loudspeaker controller that receives and processes an audio input signal and provides an audio output signal. A dynamic model of the loudspeaker receives the audio output signal, and models the behavior of the loudspeaker and provides predictive loudspeaker behavior data indicative thereof. The open loop loudspeaker controller receives the predictive loudspeaker behavior data and the audio input signal, and provides the audio output signal as a function of the audio input signal and the predictive loudspeaker behavior data.

Abstract: A method and system for implementing an acoustical front end customization are disclosed. The customization is implemented to optimize the sound level for each individual cochlear implant user. A known audio signal is generated using a sound source and captured by a microphone system. The captured sound signal is sampled at one or more locations along the signal processing pathway, and a spectrum is determined for the sampled signal and the known signal. A ratio of the two spectrums is related to the undesired transformation of the sampled signal, and a digital filter is designed based on the ratio to filter out the undesired transformation.

Abstract: According to one embodiment, an acoustic correction apparatus includes an input module, a calculator, a divider, a converter, an extractor, a synthesizer, and a generator. The input module receives an audio signal propagated through a sound field. The calculator calculates an impulse response from the audio signal. The divider divides the impulse response into first and second impulse responses. The converter converts the first and second impulse responses into first and second frequency spectrums. The extractor specifies an amplitude component of the first frequency spectrum with a peak relatively higher than that of the amplitude component of the first frequency spectrum, and extracts the peak as a resonance component. The synthesizer synthesizes a first property and a second property for attenuating the resonance component. The generator generates a correction filter for performing correction to obtain the synthesized property.

Abstract: An active noise reduction system that reduces the incidence of divergence in the presence of high amplitude interfering noise. A limited frequency range threshold is established.

Abstract: In a simulation apparatus, a difference data generation part compares a simulation result of an existing sound generator with a simulation result of a virtual sound generator, and generates difference data representing a difference between the simulation result of the existing sound generator and the simulation result of the virtual sound generator. A characteristic correction part corrects a measurement result of the existing sound generator based on the difference data, and generates virtual sound generator prediction data representing the sound generation characteristic of the virtual sound generator according to the corrected measurement result of the existing sound generator.

Abstract: A method of operating a loudspeaker includes providing a digital audio signal and identifying a target transfer function to be applied to the signal. At least one coefficient of an FIR filter is generated. The generating includes performing Heyser spiral curve fitting, and fitting a three-dimensional curve based on a magnitude and phase of a target transfer function. The digital audio signal is filtered through the FIR filter. The filtered signal is inputted into the loudspeaker.

Abstract: Disclosed herein are apparatus, method, and computer program product whereby a device receives an acoustic signal. In response to the received acoustic signal, the device outputs electrical signals from a first input audio transducer and a second input audio transducer. The second input audio transducer is less sensitive than the first input audio transducer.

Abstract: An active noise reduction system using adaptive filters. A method of operation the active noise reduction system includes smoothing a stream of leakage factors. The frequency of a noise reduction signal may be related to the engine speed of an engine associated with the system within which the active noise reduction system is operated. The engine speed signal may be a high latency signal and may be obtained by the active noise reduction system over audio entertainment circuitry.

Abstract: A system and method are provided for smoothing the in-room acoustic magnitude response of an audio reproduction system. An in-room acoustic magnitude response analysis is performed to determine a room resonance induced peak associated with an audio signal. A replica of the audio signal is filtered at the room resonance induced peak. The filtered replica signal is added with the audio signal. Through this, smoothing of the room resonance induced peak may be achieved, such that a subjective impression of transient response and dynamics of the audio signal are preserved.

Abstract: To reliably and consistently detect desirable sounds, a system detects the presence of wind noise based on the power levels of audio signals. A first transducer detects sound originating from a first direction and a second transducer detects sound originating from a second direction. The power levels of the sound are compared. When the power level of the sound received from the second transducer is less than the power level of the sound received from the first transducer by a predetermined value, wind noise may be present. A signal processor may generate an output from one or a combination of the audio signals, based on a wind noise detection.

Abstract: An audio device includes a first audio path with a loudspeaker for reproducing an audio signal, and a second audio path. The second audio path includes in series a band-pass filter for filtering an audio signal, a detector for detecting the amplitude of the band-pass filtered audio signal, a multiplier for multiplying a periodic signal by the amplitude of the band-pass filtered audio signal, and a vibration device for reproducing the multiplied periodic signal. The frequency of the periodic signal is substantially equal to the resonance frequency of the vibration device.

Abstract: Methods of acoustic dose monitoring and acoustic dose systems are provided. An acoustic dose system includes a first device configured to measure sound pressure levels directed to an ear drum and a second device configured to convert the sound pressure levels into a sound pressure level dose. The second device sends a notification signal to a third device if the sound pressure level dose is larger than a threshold value.

Abstract: There is provided a microphone that can suppress vibration noise stemming from mechanical vibrations and that outputs a collective signal having superior quality. There is provided a microphone comprising a first capacitance section M1 having a first movable electrode 101 and a second electrode 102 disposed opposite the first electrode 101 and a second capacitance section M2 that has a first movable electrode 111 and a second electrode 112 disposed opposite the first electrode 111; a first amplifier 201 that amplifies a signal from the first electrode 101 of the first capacitance section M1 and a signal from the second electrode 112 of the second capacitance section M2; and a second amplifier 202 that amplifies a signal from the second electrode 102 of the first capacitance section M1 and a signal from the first electrode 111 of the second capacitance section M2.

Abstract: A speaker array and microphone arrays positioned on both sides of the speaker array are provided. A plurality of focal points each serving as a position of a talker are set in front of the microphone arrays respectively symmetrically with respect to a centerline of the speaker array, and a bundle of sound collecting beams is output toward the focal points. Difference values between sound collecting beams directed toward the focal points that are symmetrical with respect to the centerline are calculated to cancel sound components that detour from the speaker array to microphones. Then, it is estimated based on totals of squares of peak values of the difference values for a particular time period that the position of the talker is close to which one of the focal points, and the position of the talker is decided by comparing the totals of the squares of the peak values of the sound collecting beams directed to the focal points that are symmetrical mutually.

Abstract: A system for enhancing the sound signal produced by an audio system in a listening environment by compensating for ambient noise in the listening environment is provided. The system receives an electrical sound signal and generates a sound output therefrom. A total sound signal is sensed representative of the total sound level in the environment, where the total sound level includes both the sound output from the audio system and the ambient noise within the environment. The system extracts an ambient noise signal representative of the ambient noise in the environment from the total sound signal in response to the total sound signal and to a reference signal derived from the electrical sound signal. The system extracts the ambient noise signal using an adaptive filter with an adaptive step size. The system generates a control signal in response to the ambient noise signal and adjusts the sound output of the audio system to compensate for the ambient noise level in response to the control signal.

Abstract: When the frequency of an engine rotation signal reaches a predetermined frequency, a comparator of a switching unit outputs a switching control signal to selectors and a filter coefficient updater. Based on the switching control signal, the selector switches from a connection between one memory and a corrector to a connection between another memory and the corrector, thereby changing the transfer characteristics C^rr of the corrector from C^11 to C^10. Based on the switching control signal, the selector switches from a connection between one ADC and a filter coefficient updater to the connection between another ADC and the filter coefficient updater, thereby supplying the filter coefficient updater with an error signal, rather than an error signal.

Abstract: In a communication machine room wideband noise suppression system, a sensing unit detects a noise source produced by a fan during the operation thereof and generates a noise input signal and a feedback signal, which are sent to a signal amplifying unit for signal amplification. The amplified signals are then sent to a signal converting unit and converted into digital signals. A multi-channel hybrid controller receives the digital signals and makes corrections and conduct rapid convergence algorithm to derive a reverse digital signal, which is sent to the signal converting unit and converted into a reverse analog signal. The reverse analog signal is sent to the signal amplifying unit for power amplification to generate a control signal for driving a loudspeaker unit to produce interfering acoustic wave, so as to cancel out the noise source and achieve the purpose of eliminating wideband noise.

Abstract: When an inputted audio signal is outputted from a speaker device having a predetermined input-output characteristic, the predetermined input-output characteristic being selected such that the linearity of an output level with respect to an input signal is approximately assured at a level equal to a predetermined level or more and the output level with respect to the input signal is lowered at a level equal to the predetermined level or less, a correction process for compensating a lowered output level is carried out with respect to a signal component approximately of a level equal to the predetermined level or less in the inputted audio signal. Owing to the correction process, the reproduction characteristic of a small volume signal from a speaker is improved.

Abstract: An audio output apparatus includes: an audio codec outputting an analog audio signal corresponding to a digital audio signal from a system controller; a switch unit having a first end coupled to the audio codec through a capacitor, and a grounded second end; and a switch controller triggered by a trigger signal to output a control signal to a control end of the switch unit such that the switch unit couples the capacitor to ground in response to the control signal. The trigger signal is generated by one of the system controller, the audio codec, and a power circuit supplying electric power to the system controller, the audio codec and the switch controller upon occurrence of a condition associated with pop noise, and is outputted to the switch controller before the pop noise is generated, such that the pop noise is conducted to ground via the switch unit.

Abstract: Methods and apparatuses for media playback system are disclosed, including a signal source for providing electrical signal representing media content; a processing unit for processing signal from the signal source; a distorting unit for introducing distortion to signal from the processing unit; a transducer for converting the electrical signal from the distorting unit to energy representing the media content; and a connection link for transmitting the signal from the distorting unit to the transducer. The arrangement protects media content against unauthorized duplication by ripping-off the transmitting signal.

Abstract: A method and system for implementing an acoustical front end customization are disclosed. The customization is implemented to optimize the sound level for each individual cochlear implant user. A known audio signal is generated using a sound source and captured by a microphone system. The captured sound signal is sampled at one or more locations along the signal processing pathway, and a spectrum is determined for the sampled signal and the known signal. A ratio of the two spectrums is related to the undesired transformation of the sampled signal, and a digital filter is designed based on the ratio to filter out the undesired transformation.

Abstract: A compression circuit of a device providing feedforward-based ANR monitors the electric signal output by a feedforward microphone for indications of the voltage levels of the electric signal output by the feedforward microphone ceasing to have a linear relationship with the acoustic levels of the sounds detected by the feedforward microphone. As long as there are no such indications, the compression circuit relays a signal to a feedforward anti-noise generator that is at least representative of the electric signal output by the feedforward microphone in which the sounds represented are not compressed, perhaps by directly relaying the signal output by the feedforward microphone as feedforward reference sounds.

Abstract: A method and apparatus for loudspeaker equalization are described. In one embodiment, the method comprising generating a set of parameters using an invertible, non-linear system based on input audio data and output data corresponding to a prediction of an output of a loudspeaker in response to the input data, and controlling an exact non-linear inverse of the non-linear system using the set of parameters to output a predistorted version of the input data.

Abstract: An active noise reduction system that reduces the incidence of divergence in the presence of high amplitude interfering noise. A limited frequency range threshold is established.

Abstract: A method of determining a signal input to a transducer from the signal output from the transducer, the method comprising receiving as input at a processing means the signal output from the transducer, processing the signal output from the transducer in dependance upon a value for the linear gain coefficient of the transducer and a value for the quadratic nonlinear coefficient of the transducer, to determine the signal input to the transducer.

Abstract: Neural networks provide efficient, robust and precise filtering techniques for compensating linear and non-linear distortion of an audio transducer such as a speaker, amplified broadcast antenna or perhaps a microphone. These techniques include both a method of characterizing the audio transducer to compute the inverse transfer functions and a method of implementing those inverse transfer functions for reproduction. The inverse transfer functions are preferably extracted using time domain calculations such as provided by linear and non-linear neural networks, which more accurately represent the properties of audio signals and the audio transducer than conventional frequency domain or modeling based approaches. Although the preferred approach is to compensate for both linear and non-linear distortion, the neural network filtering techniques may be applied independently.

Abstract: In an active noise cancellation system having an adaptive filter that outputs a control signal, first and second speakers that emit a canceling signal generated based on the control signal, a microphone that detects an error signal, a correction filter that corrects the base signal by a correction value to generate a reference signal and a filter coefficient updater that successively updates the adaptive filter coefficient based on the error signal and reference signal such that the error signal is minimized, the correction value of the correction filter is set to a sum obtained by adding the transfer characteristic from the first speaker to the microphone, and a product obtained by multiplying the transfer characteristic from the second speaker to the microphone by the prescribed value, thereby enabling to reduce the number of microphones and avoid the increase in parts, the amount of work to provide complicated wiring to the microphones, and the computational load involved in updating the adaptive filter co

Abstract: To reliably and consistently detect desirable sounds, a system detects the presence of wind noise based on the power levels of audio signals. A first transducer detects sound originating from a first direction and a second transducer detects sound originating from a second direction. The power levels of the sound are compared. When the power level of the sound received from the second transducer is less than the power level of the sound received from the first transducer by a predetermined value, wind noise may be present. A signal processor may generate an output from one or a combination of the audio signals, based on a wind noise detection.

Abstract: The present invention provides a cancellation signal which may be combined with a radio or television program signal to provide a mix minus signal for use in IFB communication with the talent. A cancellation signal circuit is responsive to a talent signal to delay and gain adjust the talent signal to provide the cancellation signal. The amount of gain and delay necessary may be adjusted by an operator, or automatically determined or set in response to the program signal or the mix minus signal. Automatic operation is provided even in the presence of changing delay and amplitude of the program signal or the talent signal component thereof.

Abstract: An audio device, for example a wireless communications handset, including a sound transducer (410) coupled to a compensated audio signal output of an audio compensator (450), a mismatch detection circuit (430) having a first input coupled to the compensated audio signal output of the audio compensator (450), the mismatch detection circuit (430) having a second input coupled to the sound transducer (410), the mismatch detecting circuit having an output corresponding to a mismatch between a reference electrical impedance of the sound transducer and an actual electrical impedance of the sound transducer, a compensation estimator (440) having an input coupled to the output of the mismatch detection circuit, the compensation estimator having an audio compensation output coupled to a compensation input of the audio compensator.

Abstract: An active sound muffler for reducing a sound to be reduced as emitted from a sound source located at one of the opposite sides of a sound insulating wall and diffracted and transmitted to the other side, the muffler comprises a control loudspeaker arranged at the front end or the other side of the sound insulating wall and adapted to output a control sound with a predetermined amplitude and a predetermined phase, a control microphone arranged above the sound insulating wall and adapted to gauge the sound pressure or the acoustic intensity of the sound to be reduced and that of the control sound and a control circuit for controlling the output of the control loudspeaker so as to minimize the sound pressure or the acoustic intensity, whichever appropriate, based on the outcome of gauging of the control microphone, and the control loudspeaker showing a line sound source characteristic.

Abstract: The invention is a variable damping circuit for a loudspeaker that provides for frequency dependent control of a loudspeaker by an amplifier. The variable damping circuit includes an amplifier, at least one loudspeaker, a pair of connectors between the amplifier and the loudspeaker, and a reactive component wired in parallel with a resistor connected to one of the pair of connectors between the amplifier and loudspeaker, wherein the resistance value of the reactive component and resistor does not exceed fifty per cent of a resistance value of the loudspeaker. The reactive component may be a coil, a capacitor, or both a coil and a capacitor both of which are connected in parallel with the resistor.

Abstract: A system and method for actively damping boom noise within an enclosure such as an automobile cabin. The system comprises an acoustic wave sensor, a motion sensor, an acoustic wave actuator, a first electronic feedback loop, and a second electronic feedback loop. The enclosure defines a plurality of low-frequency acoustic modes that can be induced/excited by the enclosure cavity, by the structural vibration of a panel of the enclosure, by idle engine firings, and a combination thereof. The acoustic wave actuator is substantially collocated with the acoustic wave sensor within the enclosure. The motion sensor can be secured to a panel of the enclosure.

Abstract: In a device having a sound source within an enclosure and a first microphone for picking up sounds outside the enclosure, a system for improving acoustic separation between the sound source and first microphone comprising an additional microphone located approximately equidistant from the sound as the first microphone for picking up sounds within the enclosure, and a circuit connected to the first microphone and the additional microphone for subtracting the sounds within the enclosure from the sounds outside the enclosure.