Abstract: A noise reduction system includes multiple transducers that generate time domain signals. A transforming device transforms the time domain signals into frequency domain signals. A signal mixing device mixes the frequency domain signals according to a mixing ratio. Frequency domain signals are rotated in phase to generate phase rotated signals. A post-processing device attenuates portions of the output based on coherency levels of the signals.

Abstract: A switch system and novel method control the application of noise reduction upon a signal.

Abstract: A mobile wireless communications device includes a housing and circuit board carried by the housing and having radio frequency (RF) circuitry and a processor operative with each other. Audio circuitry is carried by the circuit board and an audio transducer assembly, such as a speaker, has electrical contacts that electrically engage the audio circuitry for carrying audio signals between the audio circuitry and audio transducer assembly. A filter is mounted at the audio transducer assembly and reduces radio frequency (RF) electromagnetic interference to the audio transducer assembly during device operation.

Abstract: In one embodiment, a two-way telecommunication device may perform acoustic echo cancellation on incoming signals. An audio decoding module may produce an audio render signal. An audio capture interface may receive an audio capture signal. A short length adaptive filter may determine a time delay between the audio render signal and the audio capture signal by adaptively predicting a sub-band of the audio capture signal using a corresponding sub-band of the audio render signal.

Abstract: An audio enhancement system for compensating for ambient noise in a listening environment, comprises an audio system that produces an electrical sound signal and generates a sound output from the electrical sound signal. A sensor (e.g., a microphone) senses a total sound signal representative of the total sound level in the listening environment, including the sound output from the audio system and the ambient noise within the listening environment. A processing unit responsive to the total sound signal and the electrical sound signal extracts from the total sound signal an ambient noise signal representative of the ambient noise in the listening environment. A controller responsive to the ambient noise signal performs a linear predictive coding (LPC) analysis and generates a control signal, which is input to an equalizer to adjust the sound output of the audio system in order to compensate for the ambient noise level.

Abstract: An audio output apparatus includes a masking band determining unit configured to determine a first frequency band in which masking due to environmental sounds is likely to occur in audio signal output sounds; a band-component extracting unit configured to extract a signal component from an input audio signal in the first frequency band determined by the masking band determining unit; a pitch shift unit configured to perform pitch shifting of the signal component in the first frequency band extracted by the band-component extracting unit and generate a pitch shift signal containing a signal component of at least a doubled frequency; and a signal output unit configured to supply an audio signal containing the pitch shift signal acquired by the pitch shift unit to a connected speaker.

Abstract: Methods and systems for masking audio noise are disclosed. One apparatus includes a silence detector configured to detect a period of substantial silence in an audio signal; a masking noise source operably coupled to the silence detector, the masking noise source configured to generate a noise signal in response to the silence detector detecting the period of substantial silence; and at least one combining device operably coupled to the masking noise source, the at least one combining device configured to contribute to combining the audio signal and the noise signal. A method includes detecting a period of substantial silence in an audio signal; and combining masking noise with the audio signal during the period of substantial silence.

Abstract: A radio configured to dynamically control cancellation of undesired signals in an audio stream. The radio includes a noise cancellation processor configured to receive an audio stream from a user and to alter information in the audio stream by filtering out undesired signals in the audio stream. The radio also includes a receiving component configured to receive a data packet from a remote device, to retrieve configuration information from the data packet, and to dynamically apply the configuration information, while the radio is being used by a user, to settings associated with the noise cancellation processor. A dynamically enabled noise cancellation processor suppresses undesired signals associated with a subsequent incoming audio stream provided by the user and transmits at least one of an altered audio stream or an unaltered audio stream to the remote device.

Abstract: Noise discrimination in signals from a plurality of sensors is conducted by enhancing the phase difference in the signals such that off-axis pick-up is suppressed while on-axis pick-up is enhanced. Alternatively, attenuation/expansion are applied to the signals in a phase difference dependent manner, consistent with suppression of off-axis pick-up and on-axis enhancement. Nulls between sensitivity lobes are widened, effectively narrowing the sensitivity lobes and improving directionality and noise discrimination.

Abstract: A system for monitoring the operation of one or more devices is provided. The system includes a microphone acoustically coupled to one the monitored device. An analog-to-digital-converter samples the microphone and a processor examines the resultant digital signal for the occurrence of an abnormal event.

Abstract: The present invention provides a novel system and method for monitoring the audio signals, analyze selected audio signal components, compare the results of analysis with a threshold value, and enable or disable noise reduction capability of a communication device.

Abstract: A method for sound source localization comprises the steps of computing an envelope of a signal; performing a non-linear, onset preserving smoothing of the envelope; and evaluating cues for sound source localization, using the points where the smoothing changes from its rising phase to its falling phase as measurement points.

Abstract: An audio output device and method capable of preventing generation of pop noise by a relatively simple and small-sized circuit configuration and control timing. A D/A converter converts a digital audio signal to an analog audio signal. A non-inverting amplifier amplifies the analog audio signal. A signal output from the non-inverting amplifier is amplified by inverting amplifiers of two stages with a first timing. A signal output from an inverting amplifier is input to a speaker amplifier by a switch. The output signal of the inverting amplifier is output to a speaker with a second timing subsequent to the first timing. The output signal of the inverting amplifier is amplified by the speaker amplifier and output to the speaker. With a third timing subsequent to the second timing, the output of each of the inverting amplifiers is stopped and the output of the speaker amplifier is stopped.

Abstract: Methods and apparatuses for deriving a signal-to-noise ratio based at least in part on a measured level of a signal carrying far-end speech, and a measured level of a signal carrying ambient acoustic noise, determining a target gain adjustment based at least in part on the derived signal-to-noise ratio, including determining a coarse gain adjustment and a fine gain adjustment, the target gain adjustment corresponding to a combination of the coarse gain adjustment and the fine gain adjustment, applying the coarse gain adjustment to the signal carrying far-end speech using first gain adjustment circuitry to produce a first gain-adjusted signal, applying the fine gain adjustment to the signal carrying far-end speech using second gain adjustment circuitry to produce a second gain-adjusted signal, and providing a result of combining the first gain-adjusted signal and the second gain-adjusted signal to audio output from a wireless communications device.

Abstract: A microphone includes a sensing element having two opposing sides; and a housing including a first acoustic port having an external-facing portion defined in part by a first aperture located on a first housing side and an internal-facing portion defined in part by a first cavity within the housing, the first cavity being coupled to a first side of the element; and a second acoustic port having an external-facing portion defined in part by a second aperture located on the first housing side and an internal-facing portion defined in part by a second cavity within the housing, the second cavity being coupled to a second side of the element. The ports are spaced apart at a distance such that a level of an electrical response by the element to an ambient acoustic noise at 50 dB A-weighted sound pressure level exceeds an internal electrical noise level of the element.

Abstract: It is an object of the present invention to provide a howling suppressing apparatus and an acoustic apparatus which can suppress the howling component effectively without suppressing a periodic component other than the howling component. The howling control apparatus comprises a signal separating unit 23 operable to separate an input signal into two components including a periodic component and a nonperiodic component, a howling detecting unit 24 operable to detect a howling component from the input signal on the basis of signals selected from among the input signal and periodic and nonperiodic component signals from the signal separating unit 23, a howling suppressing unit 25 operable to suppress the howling component of the input signal on the basis of detection information from the howling detecting unit 24, and a signal synthesizing unit 26 operable to synthesize a signal from signals from the howling suppressing unit 25.

Abstract: A sound quality correcting apparatus includes: an input module; a feature quantity calculator; a score calculator; a modulation spectrum power calculator; a score corrector; and a signal corrector. The input module receives an input audio signal. The feature quantity calculator calculates feature quantities of the input audio signal for each of a plurality of first intervals having a certain time length. The score calculator calculates a score value for each the first interval based on the feature quantities. The modulation spectrum power calculator calculates a power value, at a certain modulation frequency, of a modulation spectrum of the input audio signal. The score corrector corrects score values in the plurality of first intervals that belong to a second interval if a power value calculated in the second interval is larger than or equal to a certain value. The signal corrector corrects the audio signal based on the score values.

Abstract: An amplifier circuit (100) has an input stage (OP1) and an output stage (Q1, Q2) operating with different supply voltages and different quiescent voltages. The output stage has a feedback input connected to receive a feedback signal from the output of the output stage. A biasing circuit (602) applies a bias signal (Ioff) to said input stage at an operating level appropriate to establish a quiescent output voltage different from a ground reference level of the input stage.

Abstract: The invention concerns a method and signal conversion device for avoiding undesirable noise in the start up of an amplifying device, as well as to an amplifying device including such a signal conversion device. The signal conversion device (12) comprises a variable gain providing unit (Q3, Q4, Q5, Q6), a voltage to current converter (Q1, Q2, R1, R2), a variable gain control unit (22) controlling the variable gain of the variable gain providing unit, and a bias current control unit (20) for controlling a first biasing current (IB1) of the voltage to current converter, for avoiding DC offset originating noise as well as noise originating from components of the signal conversion device.

Abstract: Various embodiments for components and associated methods that can be used in a binaural speech enhancement system are described. The components can be used, for example, as a pre-processor for a hearing instrument and provide binaural output signals based on binaural sets of spatially distinct input signals that include one or more input signals. The binaural signal processing can be performed by at least one of a binaural spatial noise reduction unit and a perceptual binaural speech enhancement unit. The binaural spatial noise reduction unit performs noise reduction while preferably preserving the binaural cues of the sound sources. The perceptual binaural speech enhancement unit is based on auditory scene analysis and uses acoustic cues to segregate speech components from noise components in the input signals and to enhance the speech components in the binaural output signals.

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: An equalization method using equal loudness curve, and a sound output apparatus using the same are disclosed. An audio signal is equalized using a standard loudness curve so as to increase a loudness of an audio signal. A sound quality can be significantly enhanced with the enhancement of a sound cleanness and quality when being adapted to a communication apparatus such as a telephone terminal, a transmitter or a repeater. When it is adapted to an audio apparatus such as a MP3, a common audio system, a car audio system, etc., it is possible to enhance a sound cleanness and quality, and a hearing ability. It can be used as a hearing aid.

Abstract: A microphone system has a base coupled with first and second microphone apparatuses. The first microphone apparatus is capable of producing a first output signal having a noise component, while the second microphone apparatus is capable of producing a second output signal. The system also has combining logic operatively coupled with the first microphone apparatus and the second microphone apparatus. The combining logic uses the second output signal to remove at least a portion of the noise component from the first output signal.

Abstract: A system comprising converter logic that introduces noise to a signal. The system also comprises amplifier logic that reduces a sound pressure associated with the noise by amplifying the signal prior to providing the signal to the converter logic and de-amplifying the signal after providing the signal to the converter logic.

Abstract: Particular embodiments provide for attenuating one or more microphone signals in a teleconferencing system upon detecting a non-audible signal. A far end voice signal is received from a sound source. A non-audible signal is added to the far end voice signal to create a composite signal, which is provided to one or more speakers. The speakers output the composite signal and the non-audible signal are detected in the composite signal after the composite signal is received at one or more microphones. The non-audible signal allows an attenuator to attenuate a microphone signal including the composite signal from a particular microphone in response to the detected non-audible signal to reduce far end echo.

Abstract: Speakers deployed in a space and divided into groups associated with different zones produce a mix of sounds that create internal noise pollution and, combined with external noise, an unpleasant environment for occupants. The present invention contemplates sound systems and methods for creating personalized sound zones to address these and related problems.

Abstract: A compact noise suppression circuit for a small speaker system and having a circuit board with a left signal input connection and a right signal input connection and a ground connection, and further having a left speaker supply connection and a left speaker return connection, and a right speaker supply connection and a right speaker return connection, and further having a left bifilar coil and a right bifilar coil, and left and right coil connections on the circuit board, and means securing said left and right bifilar coils to respective coil connections on the circuit board.

Abstract: Disclosed herein are system, method and apparatus with environmental noise cancellation. The instant disclosure is particularly adapted to a receiver module having at least two inputs. The two inputs respectively receive a main audio portion and the audio with majority of environmental noise. The system firstly calibrates the audio signals to reduce the error caused by the difference between the two inputs. An adaptive beamforming technology and a speech extractor are respectively used to extract the environmental noise portion with less main audio and the main audio portion with less noise. After a process of time-to-frequency domain transformation, a non-linear noise suppression technology is introduced into estimating the environmental noise and acquiring a gain. After noise suppression processed with the gain, a sequence of audio signals is output after a frequency-to-time domain transformation.

Abstract: A noise reduction system includes multiple transducers that generate time domain signals. A transforming device transforms the time domain signals into frequency domain signals. A signal mixing device mixes the frequency domain signals according to a mixing ratio. Frequency domain signals are rotated in phase to generate phase rotated signals. A post-processing device attenuates portions of the output based on coherence levels of the signals.

Abstract: A microphone for use in windy environments. The microphone uses a transducer to flood the environment outside the microphone with a high frequency (such as an ultrasonic) acoustic field. The sounds desired to be detected are mixed with the high frequency carrier, and can then be received by the microphone with less wind noise. The microphone then demodulates the desired sound signals from the high frequency carrier. The microphone can be configured in a special emitter-receiver configuration that also reduces interference from engine noise.

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: A signal processing device includes a noise analysis unit for analyzing a frequency component of a noise signal obtained by converting a collected sound into an electrical signal, a plurality of filtering units for carrying out predetermined filtering operations on the noise signal on the basis of an analysis result, and an output control unit for temporally varying a synthesis rate of outputs of the plurality of filtering units according to a change in the analysis result of the noise analysis unit. When the analysis result of the noise analysis unit changes, one filtering unit starts a predetermined filtering operation by characteristics different from those of other filtering units that carry out predetermined filtering operations on the noise signal according to the change in the analysis result of the noise analysis unit.

Abstract: Methods and apparatus to monitor audio/visual content from various sources are disclosed. An example method to monitor a media content presentation comprises determining whether an audio signal is muted based on whether a number of zero crossings exceeds a threshold, and based on whether the audio signal is determined to be muted, using the audio signal to at least one of determine a source of the media content presentation or determine a special operating mode of the source of the media content presentation. Another example method to monitor a media content presentation comprises determining the source of the media content presentation based on whether the audio signal is determined to be compressed. Yet another example method to monitor a media content presentation comprises determining the source of the media content presentation based on whether the video signal is determined to be compressed.

Abstract: An apparatus for acoustic beamforming comprises a beamform processor (105) for generating a beamformed signal from two audio inputs. An update processor (107) updates the beamforming filter of the beamform processor (105) if an update criterion is met. An adaptive filter (111) filters the signal from one of the signals and the difference signal between the filtered signal and the signal from the other audio input (101) is generated. An adaptation processor (115) adapts the adaptive filter (111) to minimize the difference signal. A criterion processor (109) modifies the update criterion in response to the (possibly normalized) difference signal. Specifically, the update criterion may be relaxed to improve acquisition performance if the difference signal is indicative of a strong signal outside the beam of the beamform processor (105).

Abstract: A system and method to detect and remediate unacceptable levels of speech intelligibility evaluates received test audio transmitted across and received in a space or region of interest. Intelligibility is improved by altering the rate, pitch, amplitude and frequency bands energy during presentation of the speech signal.

Abstract: An audio reproducing apparatus includes a power supply, an amplifier, a speaker, and a controller. The power supply is for supplying a voltage. The amplifier is for receiving the voltage and audio signals, amplifying the audio signals, and outputting amplified audio signals. The speaker is for reproducing sound after receiving the amplified audio signals. The controller is for receiving the voltage and generating a control signal to enable the amplifier. The controller includes a generator and a delay unit. The generator is for receiving the voltage and generating the control signal. The delay unit is for delaying the time of transferring the voltage from the power supply to the generator.

Abstract: Disclosed herein are detectors of audio ringing feedback, that is decaying feedback with a gain of less than one, those detectors utilizing a repeated gain measurement that applied to a range of gain values characteristic of ringing-type feedback. Those gain measurements, while in the range, increase a probability measurement of feedback. When the probability of feedback reaches a threshold, a detection of feedback is made and feedback countermeasures, such as the application of a notch filter, may be applied. Optionally, the audio gain around likely frequencies of feedback may be enhanced for a time to increase the resolution of identification of a feedback frequency, which may be identified through an interpolative method. Repeated gain measurements may also identify building-type feedback. A ringing detector may include more than one range of detection, for example for building, strong-ringing and weak-ringing feedback.

Abstract: A system and method are disclosed for enhancing audio signals by nonlinear spectral operations. Successive portions of the audio signal are processed using a subband filter bank. A nonlinear modification is applied to the output of the subband filter bank for each successive portion of the audio signal to generate a modified subband filter bank output for each successive portion. The modified subband filter bank output for each successive portion is processed using an appropriate synthesis subband filter bank to construct a modified time-domain audio signal. High modulation frequency portions of the audio signal may be emphasized or de-emphasized, as desired. The modification may be applied within one or more frequency bands.

Abstract: A technique for reducing inherent noise in a real time digital audio wireless system. The technique allows for low power stereo analog to digital and digital to analog converters to be used in a battery operated real time digital audio wireless applications. The audio signal is sent to the left and right channels of an analog to digital converter with a preceding audio amp of different gain at the input to each channel. The signals are simultaneously digitized then combined to generate a new digital value resulting in noise reduction at lower amplitudes. This new data word is then transmitted via radio waves to a digital audio receiver. Once the data is received it is output simultaneously to a stereo digital to analog converter. The digital to analog conversion of both left and right channels is then summed using a resistor network and an audio amplifier. The audio amplifier output is then reduced to create further reduction of noise at all amplitudes.

Abstract: The present invention provides a leakage detector enabling accurate and stable detection for occurrence and a position of leakage in a buried fluid pipe line and also making it possible for even those not so skilled in the leakage detection work to use the detector. The leakage detector 1 according to the present invention comprises a vibration detector 2 having a pickup 6 incorporating a piezoelectric element; a main body 4 of the detector incorporating voltage amplifiers 31, 37 for voltage-amplifying an output signal and a plurality types of noise removing units 33, 34, 35 for removing noises from the output signal, and a headphone 5. The main body 4 of the detector has a display unit 15 for displaying data for detected vibration sounds on a predefined screen.

Abstract: Spatial noise suppression for audio signals involves generating a ratio of powers of difference and sum signals of audio signals from two microphones and then performing noise suppression processing, e.g., on the sum signal where the suppression is limited based on the power ratio. In certain embodiments, at least one of the signal powers is filtered (e.g., the sum signal power is equalized) prior to generating the power ratio. In a subband implementation, sum and difference signal powers and corresponding the power ratio are generated for different audio signal subbands, and the noise suppression processing is performed independently for each different subband based on the corresponding subband power ratio, where the amount of suppression is derived independently for each subband from the corresponding subband power ratio. In an adaptive filtering implementation, at least one of the audio signals can be adaptively filtered to allow for array self-calibration and modal-angle variability.

Abstract: A system and method to detect and measure remediated speech intelligibility by evaluating received test audio transmitted across and received in a space or region of interest. Remediation of the test audio may include altering the rate, pitch, amplitude and frequency bands energy during presentation of the speech signal.

Abstract: A method and system to reduce the noise floor of a communications system is disclosed. The system may be incorporated into any device that provides binary samples from a datastream, such as a cordless telephone system. The system is configured to determine a number of bits of the binary samples that are affected by noise. The system is then able to remove the noise by setting those bits to a fixed value. The fixed value may depend on whether the sample is positive or negative. The value to set may be chosen so that the least significant bits of each sample come as close as possible to 0 for that particular numerical representation system. The system can be integrated with other known signal processing methods.

Abstract: A distortion compensation system minimizes distortion in an audio system by monitoring a supply voltage and adjusting a clipping threshold and/or compression knee. An adjustable gain circuit controls the gain of the audio signal according whether the audio signal exceeds a variable threshold. The variable threshold is adjusted within a threshold range based on the supply voltage. Distortion due to clipping of the audio signal is minimized while available power at any given time is maximized.

Abstract: A feedback calibration system and a method for controlling an electronic signal are disclosed. The feedback calibration system includes an input controller adapted to modify an input signal in response to a control signal and generate a modified input signal, a signal processing block including a signal analyzer, wherein the signal processing block is adapted to process the modified input signal to generate an output signal and the signal analyzer is adapted to detect an undesirable condition of the output signal and transmit a detection signal corresponding to the undesirable condition, a transfer function estimator adapted to model and transmit a transfer function estimate of the signal processing block in real-time in response to the detection signal, and a programmable device adapted to transmit the control signal to the input controller for modifying the input signal, wherein the control signal is based upon the transfer function estimate.

Abstract: A method for manufacturing an acoustical device, especially a hearing device. A device casing is provided with an acoustical/electrical input converter arrangement with an electric output. An audio signal processing unit establishes audio signal processing of the device according to individual needs and/or purpose of the device. At least one electrical/mechanical output converter is provided. A filter arrangement with adjustable high-pass characteristic has a control input for the characteristic. The following operational connections are established: between the output of the input converter arrangement and the input of the filter arrangement, between the output of the filter arrangement and the control input, between said output of the filter arrangement and the input of the processing unit, between the output of the processing unit and the input of the at least once output converter.

Abstract: A multi-channel decoding method includes: receiving an input signal to generate a first channel output signal and a second channel output signal, wherein the input signal is mixed with a specific clock signal; and gradually changing an amplitude of the specific clock signal from a first value to a second value when switching from a first mode corresponding to a first number of channels to a second mode corresponding to a second number of channels. Systems utilizing the method and another method further comprising calibration are also disclosed.

Abstract: A signal processing apparatus includes first and second extracting units extracting frequency components having a first frequency band and a second frequency band, respectively, from an input audio signal, a first-harmonic-component generating unit generating a first-harmonic-component signal including a frequency component whose frequency is N1 times that of the frequency component extracted in the first extracting unit, a second-harmonic-component generating unit generating a second-harmonic-component signal including a frequency component whose frequency is N2 times that of the frequency component extracted in the second extracting unit, and a combining unit combining the input audio signal, and the first- and the second-harmonic-component signals in a predetermined ratio. The first frequency band is higher than the second frequency band. N1 and N2 are positive integers, and N1 is smaller than N2.

Abstract: Masking thresholds are obtained for each frequency component of sound data and ambient noise. It is determined whether each frequency component of the sound data is masked by at least one of the other frequency components of the sound data. It is further determined whether each frequency component of the sound data is masked by ambient noise. Correction coefficients are set for each frequency component of the sound data according to whether the frequency component is masked by at least one of the other frequency components of the sound data and whether the frequency component is masked by the ambient noise. And each frequency component of the sound data is corrected by using the respective correction coefficients.

Abstract: A grounding switch is described which operates properly even in the presence of negative voltages on a signal line. The grounding switch uses isolated field effect transistors that have their substrates tied to different voltages. The isolated field effect transistor has gate voltage and substrate voltage which can be pulled down to negative voltage when the signal line has a negative voltage allowing the switch to remain open even with a negative voltage.