Abstract: Systems, methods, and devices for improving speaker performance with an acoustic damped port are disclosed. In accordance with various embodiments of the present invention, a damping material is placed around a vented frame of a speaker driver, and substantially covers or fills the vents of the frame. In some embodiments, the damping material results in improved impedance matching with acoustic delay, without required dimensional tuning (as with conventional ports). In some embodiments, it also reduces the air velocity gradients minimizing higher order frequency distortion components. In some embodiments, the damping material also acts as an absorber of energy by coupling to the sound wave. In some embodiments, low-frequency performance is improved in a small-scale design.

Abstract: In a method and equipment for operating a voice-supported system, such as a communications and/or intercom/two-way intercom device in a motor vehicle, using at least one microphone and at least one loudspeaker to reproduce a signal generated by the microphone, as well as a bandpass filter configured between the microphone and the loudspeaker, a power of the signal as a function of a frequency is determined, and the bandpass filter is adjusted as a function of at least one local maximum of the power of the signal as a function of the frequency.

Abstract: Various embodiments of systems and methods for reducing audio noise are disclosed. One or more sound components such as noise and network tone can be detected based on power spectrum obtained from a time-domain signal. Results of such detection can be used to make decisions in determination of an adjustment spectrum that can be applied to the power spectrum. The adjusted spectrum can be transformed back into a time-domain signal that substantially removes undesirable noise(s) and/or accounts for known sound components such as the network tone.

Abstract: An exemplary method of dynamically adjusting an amount of noise reduction applied in an auditory prosthesis system includes dividing an audio signal presented to a patient into a plurality of analysis channels each containing a signal representative of a distinct frequency portion of the audio signal, determining an overall noise level of the signals within the analysis channels, and dynamically adjusting an amount of noise reduction applied to the signals within the analysis channels in accordance with the determined overall noise level. The dynamic adjustment of noise reduction is configured to minimize the amount of noise reduction applied to the signals within the analysis channels if the overall noise level is less than a predetermined minimum threshold. Corresponding methods and systems are also disclosed.

Abstract: An image sensing apparatus which senses the optical image of an object using a lens unit having a movable lens can reduce the influence of noise generated upon driving the lens from the first sound signal, based on the first sound signal obtained by the first microphone unit for collecting an object sound, and a second sound signal obtained by the second microphone unit for collecting noise. The second microphone unit is arranged at a position where the relative positional relationship with the generation source of noise generated upon driving the lens does not change even if the lens is moved.

Abstract: An exemplary method of reducing an effect of ambient noise within an auditory prosthesis system includes dividing an audio signal presented to an auditory prosthesis patient into a plurality of analysis channels each containing a frequency domain signal representative of a distinct frequency portion of the audio signal, determining a signal-to-noise ratio and a noise reduction gain parameter based on the signal-to-noise ratio for each of the frequency domain signals, applying noise reduction to the frequency domain signals in accordance with the determined noise reduction gain parameters to generate a noise reduced frequency domain signal corresponding to each of the analysis channels, and generating one or more stimulation parameters based on the noise reduced frequency domain signals and in accordance with at least one of a current steering stimulation strategy and an N-of-M stimulation strategy. Corresponding methods and systems are also disclosed.

Abstract: To reduce noise in an input signal that may contain speech, first an estimate of the noise level in the signal is obtained. The level of the input signal is then compared with the noise level estimate signal to determine whether speech is dominant. Less aggressive noise reduction is applied to the input signal when speech is dominant than when only noise is present.

Abstract: A microphone signal compensation apparatus includes a plurality of audio input units to respectively receive a target signal, each audio input unit of the plurality of audio input units including a microphone; a constant filter unit to selectively apply a constant filtering calibration scheme to signals output by the plurality of audio input units to compensate for a difference in at least one characteristic among the audio input units, the constant filtering calibration scheme being estimated from an average value of a ratio of a desired signal to a reference signal among the signals output by the plurality of audio input units; and a noise remover to remove noise from the signals processed by the constant filter unit, and to separate the target signal from the signals from which the noise has been removed.

Abstract: An apparatus and method reduce noise in a complex spectrum domain to extract a target signal from input signals containing noise and target speech. Noise estimation may be performed through a filter with a filter learning coefficient that is updated according to a prior-signal-to-noise ratio (prior-SNR). Also, noise estimation accuracy may be improved by using confidential weighted scores. The target signal may be extracted by representing candidates of the target signal as at least two circles in the complex spectrum domain using the estimated noise and then geometrically calculating the intersections of the circles.

Abstract: A system and method for discarding or inserting audio frames in a jitter buffer is described. The system and method provides improved audio quality as compared to conventional jitter buffer management systems. In one embodiment, buffer control logic determines whether to discard audio frames to be stored in a jitter buffer or to insert audio frames among audio frames to be output from a jitter buffer based not only on the number of audio frames currently stored in the jitter buffer but also based on the power of the current audio frame to be stored in or output from the jitter buffer. The system and method is generally applicable to any wireless or wired communication system in which audio signals are transmitted between entities operating in different clock domains.

Abstract: The method comprises the following steps in the frequency domain: a) combining signals into a noisy combined signal; b) estimating a pseudo-steady noise component; c) calculating a probability of transients being present in the noisy combined signal; d) estimating a main arrival direction of transients; e) calculating a probability of speech being present on the basis of a three-dimensional spatial criterion suitable for discriminating amongst the transients between useful speech and lateral noise; and f) selectively reducing noise by applying a variable gain specific to each frequency band and to each time frame.

Abstract: The imaging device is provided with at least one imaging part to obtain images through shooting and at least one sound-collection part to obtain audio, collecting it together with the shooting of the imaging part, and at least one display part to display images. It performs audio-correction processing of audio obtained through the sound-collection part according to the relative relationship between the direction in which the display part displays images and the direction in which the imaging part shoots.

Abstract: An earpiece of an ANR device incorporates one or more of feedforward-based ANR; feedback-based ANR; passive noise reduction (PNR) of environmental noise sounds in the environment external to the casing of the earpiece in higher audible frequencies; a controlled leak acoustically coupling the front cavity to the environment external to the casing of the ANR device where the coupling may be through another cavity that is closable to the environment external to the casing with a leaky cover; a combination of an acoustically resistive port and a mass port coupling a rear cavity to the environment external to the casing where the coupling may be through another cavity that is closable to the environment external to the casing with a leaky cover; a feedforward microphone given acoustic access to the environment external to the casing through an aperture that is overlain with a leaky cover or that is enclosed within a cavity that is acoustically coupled to the environment external to the casing with a leak.

Abstract: A mobile device has an acoustic shock prevention circuit and prevents an acoustic shock unexpectedly occurring when receiver signals and speaker signals are output through a receiver-integrated speaker. The acoustic shock prevention circuit is preferably disposed between the receiver-integrated speaker and an audio processing unit is enabled in a receiver mode and disabled in a speaker mode. When the audio processing unit outputs audio signals partly exceeding a given output range, the enabled acoustic shock prevention circuit removes the exceeded parts of the audio signals.

Abstract: A method of suppressing noise in a signal containing speech and noise to provide a noise suppressed speech signal. An estimate is made of the noise and an estimate is made of speech together with some noise. The level of the noise included in the estimate of the speech together with some noise is variable so as to include a desired amount of noise in the noise-suppressed signal.

Abstract: An audio host device has an electrical interface having a speaker contact, a microphone contact, and a reference contact. The reference contact is shared by a microphone and a speaker. The reference contact is also directly coupled to a power return plane of the audio host device. A difference amplifier is provided, having a cold input and a hot input. The hot input is coupled to the microphone contact. A variable attenuator circuit is also provided having an input coupled to receive a signal from a sense point for the reference contact, and an output coupled to the cold input of the difference amplifier. A controller has an output coupled to control the variable attenuator. Other embodiments are also described and claimed.

Abstract: Active noise control system and method for controlling an acoustic noise generated by a noise source at a listening location, in which system and method sound is picked up in the surroundings of the listening location by a sound sensor; an electrical noise signal which corresponds to the acoustic noise of the noise source is generated and filtered adaptively in accordance with control signals. The adaptively filtered noise signal is irradiated into the surroundings of the listening location by a sound reproduction device, where a secondary path transfer function extends between the sound reproduction device and sound sensor. The noise signal is filtered with a transfer function that models the secondary path transfer function. The signals which are provided by the sound sensor after first filtering serve as control signals for the adaptive filtering.

Abstract: An approach is provided for measuring, identifying, and removing at least one sinusoidal interference signal (Ak·ej(?kt+?k), Ak·ej(?·?k?t+?k)) in a noise signal (w(t), w(?·?t)). A frequency range to be measured is split into a plurality of frequency bands (?) via a Fast Fourier Transform (FFT) filter bank. For each of the frequency bands (?), an autocorrelation matrix ({circumflex over (R)}?) is determined, wherein parameters of the autocorrelation matrices ({circumflex over (R)}?) are variably adjusted based on whether the at least one sinusoidal interference signal (Ak·ej(?kt+?k), Ak·ej(?·?k?t+?k)) is to be measured, identified, or removed and further based on at least one averaging. The autocorrelation matrices ({circumflex over (R)}?) are jointly utilized for one or more of measuring, identifying, or removing the at least one sinusoidal interference signal (Ak·ej(?kt+?k), Ak·ej(?·?k?t+?k)) in the noise signal (w(t), w(?·?t)).

Abstract: A voice enhancement logic improves the perceptual quality of a processed voice. The voice enhancement system includes a noise detector and a noise attenuator. The noise detector detects a wind buffet and a continuous noise by modeling the wind buffet. The noise attenuator dampens the wind buffet to improve the intelligibility of an unvoiced, a fully voiced, or a mixed voice segment.

Abstract: A method for cancelling background noise of an audio device comprises determining characteristic values of an audio signal to construct a characteristic signal reflecting a change trend of the audio signal, multiplying the determined characteristic signal with the audio signal to construct a multiplication signal, and amplifying the multiplication signal.

Abstract: A system for detecting noise in a signal received by a microphone array and a method for detecting noise in a signal received by a microphone array is disclosed. The system also provides for the reduction of noise in a signal received by a microphone array and a method for reducing noise in a signal received by a microphone array. The signal to noise ratio in handsfree systems may be improved, particularly in handsfree systems present in a vehicular environment.

Abstract: A method for automatic speech recognition includes determining for an input signal a plurality scores representative of certainties that the input signal is associated with corresponding states of a speech recognition model, using the speech recognition model and the determined scores to compute an average signal, computing a difference value representative of a difference between the input signal and the average signal, and processing the input signal in accordance with the difference value.

Abstract: For reducing wind noise effects in a hearing instrument, a converted acoustic signal is processed in a number of frequency bands, a low frequency band of which is chosen to be a master band. A wind noise attenuation value is determined in each frequency band, based on a signal level in the frequency band concerned and on a signal level in the master band. A further wind noise reducing effect may be achieved in hearing instruments with at least two microphones where in the presence of wind noise the instrument may be switched from a directional mode to a omnidirectional mode in which an average of the output signals of the two microphones is used as signal. In single microphone hearing instruments, the microphone signal and a delayed version of this signal are used to improve wind noise detection and reduction.

Abstract: Gain characteristics depending on the frequency of a reference signal from speakers to a passenger position in a motor vehicle, i.e., gain characteristics which are an inversion of vehicle cabin sound field characteristics, are set in a first acoustic corrector. At the passenger position, a gain characteristic curve that is flat at various frequencies is achieved to prevent gain peaks and dips from occurring at the passenger position. A sound effect generated at the passenger position is made linear depending on the state of a noise source, or more specifically, a noise source caused by an accelerating action on the motor vehicle.

Abstract: Methods and apparatus for controlling the audio characteristics of a networked voice communications device (NVCD) are presented. One method presented includes receiving a settings file, extracting at least one audio control parameter from the settings file, deriving audio processing parameters based upon a value selected from the at least one audio control parameter, and controlling the audio characteristics of the networked voice communications device using the audio processing parameters and the at least one audio control parameter. A method for providing audio parameters to an NVCD is also presented which includes establishing a settings file, which includes at least one audio control parameter, receiving a request to send the settings file, and sending the settings file over a network to the networked voice communications device.

Abstract: Noise reduction for a switching amplifier system having a differential output stage and demodulator filter responsive to complementary PWM signals includes generating in-phase PWM signals and gradually adjusting their duty cycle between a low duty cycle and the full duty cycle of the complementary PWM signals, generating full duty cycle PWM signals and gradually shifting their relative phase between in-phase and out-of-phase; and in response to a turn-on signal, adjusting the in-phase PWM signals from low to full duty cycle and shifting the relative phase from in-phase to out-of-phase, and in response to a turn-off signal shifting the relative phase from out-of-phase to in-phase and adjusting the in-phase PWM signals from full to low duty cycle for maintaining balanced charge on the demodulation filter to reduce audible noise.

Abstract: A transient noise removal system removes or dampens undesired transients from speech. When the transient noise removal system receives a speech frame, the system performs a wavelet transform analysis. The speech frame may be represented by one or more wavelet coefficients across one or more wavelet levels. For a given wavelet level, the transient noise-removal system may determine a wavelet threshold. The transient noise removal system may compare the threshold corresponding to a wavelet level to the wavelet coefficients within that level. The transient noise removal system may attenuate each wavelet coefficient based on a comparison to a threshold.

Abstract: According to one embodiment, an acoustic correction apparatus includes: a signal obtaining module configured to obtain an acoustic signal from a target space including an object and an external space; a signal output module configured to output to the target space a measurement signal; a coefficient identifying module configured to identify, on the basis of a response acoustic signal, a correction coefficient of a correction filter that reduces a resonance frequency component of a resonance in the object; a filtering module configured to use the correction filter, and filter the signal provided to the object; a noise cancelling module configured to remove, on the basis of the acoustic signal, a noise component comprised in the acoustic signal from the filtered signal; and an output module configured to output the acoustic signal, from which the noise component is removed by the noise cancelling module, to the object.

Abstract: A loudspeaker system with an endfire array of three or more loudspeakers (Zn, n=3, 4, . . . N) arranged on a line. The system has a set of filters (Fn, n=3, 4, . . . N), each loudspeaker (Zn) being connected to one corresponding filter (Fn). The filters (Fn) are super resolution beamforming filters such as to provide the endfire array with a pre-designed directivity index (DI) and a pre-designed noise sensitivity (NS).

Abstract: According to this invention, mechanical noise from a noise source of a pre-identified device can be removed, while tracing a change in gain of acoustic wave data, and noise is removed by tracing a change over time of the noise source. To this end, the gain of a signal from a microphone is automatically adjusted by an Auto level control unit (ALC). A threshold according to the gain of the ALC is stored in a register. A comparator compares audio data from the ALC with the threshold, generates weighting coefficients k and 1?k based on the comparison result, and outputs them to weighting coefficient multipliers. The weighting coefficient multipliers and an adder calculate a weighted average value of data in a memory that stores weighting coefficients of old frames and the input acoustic wave data, and update the contents of the memory by the calculation result. With this update processing, the memory stores only a NOISE component.

Abstract: There is provided a noise cancellation system, comprising: a voice input, for receiving a wanted signal; a noise input for receiving a detected signal representative of ambient noise; a signal processor, for generating a noise cancellation signal for addition to the wanted signal, the signal processor having an adjustable gain; and control circuitry, for determining a relationship between levels of the wanted signal and the detected signal, and for controlling the adjustable gain on the basis of the determined relationship.

Abstract: Provided is a method of removing an earphone noise of a portable terminal includes: recognizing a connection of an earphone through an interface unit; applying power to a mike bias port in case the earphone is recognized; sensing a signal which indicating a detachment of the earphone from the interface unit during applying power to the mike bias port; and discharging the power of the mike bias port when the detachment occurs.

Abstract: A system and process for dereverberation of multi-channel audio streams is presented which uses reverberation suppression techniques. In general, the present system and process builds a frequency dependent model of the reverberation decay and uses spectral subtraction-based reverberation reduction to achieve the aforementioned suppression. This dereverberation system and process can be used to improve automatic speech recognition (ASR) results with minimal CPU overhead.

Abstract: An acoustic system that eliminates the howling that occurs when the sound outputted by the speaker feeds back to the input device. The acoustic system comprises a digital signal processor (DSP) that divides the input audio signal into different frequency bands, and reduces the audio levels for the frequency bands where howling is most likely to occur. In one embodiment, the acoustic system comprises a sound source section that generates a test tone that substantially covers the entire human audible range such that the DSP can set the filter levels according to the feedback of the test tone. In another embodiment, the sound source section stores one waveform at a given pitch and generates waveforms of other pitches based on the stored waveform. In yet another embodiment, the pitches of the generated waveforms are dispersed into four frequency bands to create a test tone that resembles a chord or a musical tone.

Abstract: A speaker characteristic correction device obtains a first speaker information of a first speaker, obtains a first sound field characteristic at an evaluation point that is obtained by using the first speaker in advance, and obtains a second speaker parameter indicating a mechanical characteristic and an electric characteristic of a second speaker. Then, the speaker characteristic correction device calculates a correction characteristic based on the first speaker information and the second speaker parameter, and calculates the second sound field characteristic by applying the correction characteristic to the first sound field characteristic. Thereby, when the speaker type is changed, it is possible to easily calculate the sound field characteristic without performing the re-measurement by installing the speaker and without performing the re-analysis by setting the analysis condition.

Abstract: A method and apparatus for processing an audio signal to enhance the perceived lower frequency content of the audio signal when played through an audio output device, includes an input configured to receive an audio input signal, a processor configured to filter the audio input signal to produce a high frequency signal and a low frequency signal, generate an enhancement signal by producing higher frequency harmonics from the low frequency signal, including a process of self convolution, and combine the high frequency signal with the enhancement signal to produce an output signal; and an output configured to receive the output signal and produce an audio output.

Abstract: Disclosed is a noise suppression device capable of better noise suppression by means of a simpler structure and with a lighter computational load. A noise suppression device (100) has a noise suppression processor (150) to estimate the required information only from the observed information, which is the required information corrupted by noise. A correlator (154) calculates the correlation of the estimation error when the state quantity, which contains the required information, of the system at time n+1 was estimated from the information until time n or time n+1 for the observed information at only time n.

Abstract: There is provided a hearing system comprising: a hearing device to be worn at or in a user's ear for supplying audio signals to said user and comprising a sound attenuation portion for attenuating ambient sound before reaching the user's ear, means for producing audio signals at a controlled level, a loudspeaker which is included in the attenuation portion and which is oriented towards the user's ear canal for providing sound corresponding to the audio signal produced by the audio signal producing means to the user's ear canal, a microphone which is included in the attenuation portion and which is oriented towards the user's ear canal for capturing audio signals from the sound provided by the loudspeaker to the user's ear canal, and a level control unit adapted to control the level of the audio signals produced by the audio signal producing means according to the audio signals captured by the microphone.

Abstract: The invention provides an amplifier circuit of a capacitor microphone of which the noise resistance against noise of a supply voltage is enhanced. In an amplifier circuit of a capacitor microphone of the invention, while a noise component of a supply voltage is applied to one inversion input terminal of an operational amplifier of an amplification portion through a parasitic capacitor existing between an external power supply wiring and an external wiring that are adjacent to each other, the problem noise component of the supply voltage is applied to the other non-inversion input terminal by capacitive coupling to an internal power supply wiring. Therefore, the noise component is cancelled at the operational amplifier.

Abstract: A technology for eliminating or reducing interference sound from a sound signal to extract target sound is provided. Interference sound is modeled using training noise, and mixed source sound is separated using the modeled interference sound. The mixed source sound is separated into target sound and interference sound using a basis matrix of the modeled interference sound.

Abstract: A method for removing periodic noise pulses from a continuous audio signal generated in a pressurized air delivery system includes the steps of: detecting, in a time-windowed segment of the continuous audio signal generated in the pressurized air delivery system, a plurality of the periodic noise pulses having a pulse period and being representable in the form of a plurality of signal components combined by convolution; deconvolving the plurality of signal components to generate a plurality of deconvolved signal components; and removing at least a portion of the periodic noise pulses from the time-windowed segment of the continuous audio signal using the deconvolved signal components.

Abstract: A method and an acoustic signal processing system for noise reduction of a binaural microphone signal are proposed. A source signal and two interfering signals input to a left and a right microphone of a binaural microphone system respectively. A left and a right microphone signal is filtered by a Wiener filter to obtain binaural output signals of the source signal. The Wiener filter is calculated as H W ? ( ? ) = 1 - S y ? ? 1 , y ? ? 1 ? ( ? ) S v ? ? 1 , v ? ? 1 ? ( ? ) + S v ? ? 2 , v ? ? 2 ? ( ? ) , wherein Sy1,y1(?) is the auto power spectral density of the sum of the interfering signals contained in the left and right microphone signal, Sv1,v1(?) is the auto power spectral density of the filtered left microphone signal and Sv2,v2 is the auto power spectral density of the filtered right microphone signal.

Abstract: The invention provides a method for determining a noise reference signal for noise compensation and/or noise reduction. A first audio signal on a first signal path and a second audio signal on a second signal path are received. The first audio signal is filtered using a first adaptive filter to obtain a first filtered audio signal. The second audio signal is filtered using a second adaptive filter to obtain a second filtered audio signal. The first and the second filtered audio signal are combined to obtain the noise reference signal. The first and the second adaptive filter are adapted such as to minimize a wanted signal component in the noise reference signal.

Abstract: A signal processing apparatus is configured to change volume level or frequency characteristics of an input signal with a limited bandwidth in a first frequency range. The apparatus includes: an information extracting unit configured to extract second frequency characteristic information from a collection signal with a limited bandwidth in a second frequency range different from the first frequency range; a frequency characteristic information extending unit configured to estimate first frequency characteristic information from the second frequency characteristic information extracted by the information extracting unit, the first frequency characteristic information including the first frequency range; and a signal correcting unit configured to change volume level or frequency characteristics of the input signal according to the first frequency characteristic information obtained by the frequency characteristic information extending unit.

Abstract: Certain aspects of the present disclosure relate to a method for speeding up the Cadzow iterative denoising algorithm as a part of the Finite Rate of Innovation (FRI) processing and for decreasing its computational complexity.

Abstract: A digital sound system suitable for a digital speaker device for directly converting analog sound by a circuit using a ?? modulator and a mismatch shaping filter circuit to output a plurality of digital signals and a plurality of speakers driven by the plurality of digital signals. A digital speaker driving device includes a ?? modulator, a post filter, s driving circuits, and a power supply circuit to supply power to the ?? modulator, the post filter and the s driving elements and the s driving circuits are adapted to s digital signal terminals.

Abstract: A signal separator, a method and computer product for determining a first output signal describing an audio content of a useful-signal source in a first microphone signal, and for determining a second output signal describing an audio content of the useful-signal source in a second microphone signal.

Abstract: A howling detector is provided which can discriminate between howling and a signal having a strong narrow-band component, thereby detecting howling with higher accuracy. The howling analyzer includes a frequency analyzing section for analyzing a frequency of a time signal, a level calculating section for calculating a level of a signal output from the frequency analyzing section, a howling detecting section for deciding whether howling occurs or not by analyzing the level having been calculated by the level calculating section, a periodic signal detecting section for deciding whether or not time progression of the level having been calculated by the level calculating section has periodicity, and a howling deciding section for finally deciding whether howling occurs or not based on decision results of the howling detecting section and the periodic signal detecting section.

Abstract: The present invention generally relates to an audio signal or gesture detection. More specifically, the invention addresses an apparatus and a method for converting an audio signal detected by microphones or a gesture detected by an image sensing device into a directional indication of the source for the user.

Abstract: A method and system enhances an acoustic signal acquired by a microphone from an acoustic source while concurrently acquiring a Doppler signal from moving parts of the acoustic source. The acoustic signal and the Doppler signal are then analyzed according to a model to generate an enhanced acoustic signal.