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 noise suppressing apparatus suppresses a noise component of a sound signal which contains the noise component and a signal component. In the apparatus, a frequency analyzing section divides the sound signal into a plurality of frames such that adjacent frames overlap with each other along a time axis, and computes a first spectrum of each frame. A noise suppressing section suppresses a noise component of the first spectrum so as to provide a second spectrum of each frame in which the noise component is suppressed. A frequency specifying section specifies a frequency of a noise component of each frame. A phase controlling section varies a phase of the noise component corresponding to the specified frequency in the second spectrum by a different variation amount each frame.

Abstract: An apparatus for processing an audio signal and method thereof applied to an audio playback system are disclosed. The apparatus comprises a decoder, an error-correcting circuit and an audio correcting module. The method for processing audio signals in accordance with the present invention decodes the audio signal to generate a decoded signal by the decoder. Then, the error-correcting circuit performs an error-correcting algorithm in the decoded signal to generate an error indication signal and an output audio signal. And the audio correcting module corrects the output audio signal to generate a corrected audio signal when the error indication signal indicates that the output audio signal has error.

Abstract: Provided is a method and apparatus for canceling noise from a sound signal input through a microphone. The method includes filtering a high-frequency signal having a frequency that is higher than a reference frequency and a low-frequency signal having a frequency that is lower than the reference frequency from input signals obtained through a microphone array, obtaining a high-frequency target signal by canceling a noise signal from the filtered high-frequency signal using a beamforming method, obtaining a low-frequency target signal by canceling a noise signal having a phase difference that is different from a phase difference of a target signal from the filtered low-frequency signal, and obtaining a sound source signal from which noise is cancelled, by synthesizing the obtained high-frequency target signal with the obtained low-frequency target 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: A condenser microphone provided with a battery compartment having a coil spring electrode is adapted to: extend a return stroke of the coil spring electrode; prevent high frequency current from penetrating into the microphone from the coil spring electrode to suppress the occurrence of noise; and prevent the coil spring electrode from generating mechanical noise. The condenser microphone includes: a body case into which a condenser microphone unit is incorporated; a battery compartment provided in the body case; a coil spring electrode that is provided at an end of the battery compartment, and pushed and compressed by an electrode of a battery by the battery being inserted therein; and a conductive cushion that is disposed within the coil spring electrode, and can contact with the coil spring electrode while being compressed together with the coil spring electrode at least when the coil spring electrode is compressed.

Abstract: A system comprises a plurality of storage drives coupled to logic. The logic implements a noise-reducing feature selected from a group consisting of a first feature that limits system performance based on a level of ambient noise, a second feature that staggers access transactions among said storage drives, a third feature that staggers spin up among the storage drives, a fourth feature that at least partially cancels noise generated by the system, a fifth feature that limits fan speed, and combinations thereof.

Abstract: When a first noise and second noise to be generated after the first noise are generated within a predetermined period, a noise reduction unit is controlled so as to execute a first noise reduction process for an audio signal in a period including the first noise and not to execute the first noise reduction process for an audio signal in a period including the second noise.

Abstract: The present invention relates to a miniature transducer assembly for a wireless portable terminal. The miniature transducer assembly of the present invention comprises a transducer unit comprising a number of voice coils for displacing a diaphragm of said transducer unit. Moreover, the transducer assembly comprises at least one de-coupling coil for avoiding interference between the transducer unit and an antenna of the wireless portable terminal. In order to reduce the dimensions of the transducer assembly at least one de-coupling coil forms an integral part of a lead-out wire from of a voice coil.

Abstract: A multi-function communications device has a processor that generates a user interface audible tone signal. The device also has a downlink digital signal processor, and an uplink digital signal processor. A mixer has an input to receive the downlink signal and another input to receive the user interface tone signal. The uplink processor has an acoustic echo canceller having an input to receive the uplink signal and another input to receive an output from the mixer. The echo canceller may reduce the amount of both the far-end user's speech and the user interface tone that may be present in the uplink signal. The mixer may be positioned within the chain of audio signal processors, or it may be positioned outside the chain. Other embodiments are also described and claimed.

Abstract: Disclosed is an apparatus and method for processing signals such as sound signals. The sound processing apparatus includes a sound signal input unit for receiving sound signals, a harmonic noise separator for separating a harmonic region and a noise region from the received sound signals, a noise restraint index determination unit for determining an optimal noise restraint index k according to a system and circumstance, and a noise restrainer for restraining the separated noise region depending on the noise restraint index k so as to output noise attenuated signals.

Abstract: A dereverberation device includes a reverberation estimation unit for estimating a later reflection component by using information on an impulse response from a signal source to an observation point, a noise estimation unit, and a mixing unit. As a result, it is possible to obtain a high-quality dereverberated signal with a small calculation amount even in a noisy environment.

Abstract: A circuit for eliminating noise includes a sound card (30) with an audio signal output, a power supply (10) for providing working voltage to the sound card, a first transistor (Q1), a second transistor (Q2), and a third transistor (Q3). The power supply has a power good pin, a PSON# pin, and a standby voltage pin. During powering on time of the sound card, the power good pin is at low level and the second transistor is therefore turned on to ground the audio signal output so as to eliminate turn-on noise. During powering down time of the sound card, the PSON# pin turns from low to high level to turn on the third transistor before the audio power for providing working voltage to the sound card is powered down, thus the first transistor turns off, and the second transistor turns on to ground the audio signal output of the sound card so as to eliminate turn-off noise.

Abstract: A method for detecting failure in a noise cancellation system is provided. The noise cancellation system is operable to at least partially cancel an unwanted noise produced by a noise source. The method includes receiving a cancellation noise signal from the noise cancellation system. The cancellation noise signal represents a cancellation noise that, when combined with a target noise, is operable to at least partially cancel the target noise. The method also includes receiving an error signal approximately representing a portion of the target noise that remains after the cancellation noise and the target noise are combined. The method also includes determining whether the signal level changed or remains unchanged for each of the received cancellation noise signal and the error signal. The method also includes assigning a failure state of the noise cancellation system based on the determination and initiating a remedial measure that corresponds with the assigned failure state.

Abstract: A signal processing apparatus includes sound collecting elements, a noise detector for detecting a level of noise in a low-frequency band of audio signals output from the sound collecting elements, a noise reduction unit for reducing the noise in the audio signals in accordance with a signal output from the noise detector, a converter for converting the audio signals output from the noise reduction unit into pieces of audio data corresponding to channels including a low-frequency channel and other channels, a low-frequency channel controller for controlling a level of the audio data corresponding to the low-frequency channel in accordance with the level of the noise detected using the noise detector, and a level controller for controlling the level of the audio data of the low-frequency channel output from the low-frequency channel controller and levels of the pieces of audio data corresponding to the other channels output from the converter.

Abstract: A method and system for detecting wind noise are adapted to determine whether two of a plurality of sound signals acquired by a plurality of sound receiving units include wind noise.

Abstract: A system comprising audio logic adapted to convert captured sound into an audio signal. The system also comprises transmission logic which causes noise to be added to the audio signal. The system further comprises processing logic adapted to at least partially remove the noise from the audio signal by subtracting a noise waveform from the audio signal to produce a result signal. The processing logic generates the noise waveform using power level information associated with the transmission logic. The transmission logic transmits the result signal to another electronic device.

Abstract: A noise processing device and its method are provided for a video/audio system having a high definition multimedia interface (HDMI). The noise processing device includes a detecting unit, a signal generating unit, and a decision unit. The noise processing method includes using the detecting unit to monitor a variation related to an audio signal and generate a detecting signal accordingly; using the signal generating unit to produce an adjustment signal according to the detecting signal; and using the decision unit to produce an output audio signal according to the audio signal and the adjustment signal. Another embodiment of the noise processing device includes a compensation tracking unit having a control unit. The compensation tracking unit produces an output audio signal according to a difference between the output audio signal itself and the audio signal and a gain of the control unit.

Abstract: A driver device for suppression audible transients of an audio amplifier includes an amplifier for receiving an audio signal and a bias circuit configured to quickly generate a voltage level for biasing the amplifier, wherein the voltage level is maintained even if the driver device is powered off. The bias circuit may include a CMOS inverter having a negative feedback that has a standby current of less than 100 nA. The bias circuit further includes a buffer for rapidly charging an external capacitor. The buffer may change to a high impedance state rapidly when the power supply is disconnected.

Abstract: A novel adaptive beamforming technique with enhanced noise suppression capability. The technique incorporates the sound-source presence probability into an adaptive blocking matrix. In one embodiment the sound-source presence probability is estimated based on the instantaneous direction of arrival of the input signals and voice activity detection. The technique guarantees robustness to steering vector errors without imposing ad hoc constraints on the adaptive filter coefficients. It can provide good suppression performance for both directional interference signals as well as isotropic ambient noise.

Abstract: A sound signal generator testing apparatus is provided. The apparatus includes a hermetic seal (2), a sound signal input unit (4), and at least a noise suppression unit (5). The sound signal input unit is for receiving sound signals transmitted from a sound signal generator (1) under testing. Each noise suppression unit includes a sound signal input unit, a low pass filter, an inverter, and an output. The sound signal input unit is for receiving sound signals. The low pass filter is for filtering the sound signals by passing the low frequencies of the sound signals that are below a predetermined value. The inverter is for inverting phases of the sound signals transmitted from the low pass filter and producing inverted sound signals. The output unit is placed behind the sound signal input unit for outputting the inverted sound signals.

Abstract: A multichannel hearing aid (20) comprises at least one frequency channel having a compressor (38) with a compression threshold at an output level below the hearing threshold and an attack time above 0.5 seconds whereby hearing of a sudden sound in a stationary sound environment is facilitated. With this compressor, the amplification of low signal levels may be increased compared to the prior art, as the compressor kicks in to generally suppress steady noises. The gain may generally be increased as high as feasible in view of the microphone baseline noise, which should preferably be kept below the hearing threshold. Thus the user of the hearing aid will generally have the option of a higher gain of low level sounds than generally feasible with prior art hearing aids.

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: An output circuit for audio codec chip includes an audio amplifier chip and a noise eliminating circuit electrically connected to the audio codec chip and the audio amplifier chip. The audio amplifier chip receives audio signals form the audio codec chip and outputs amplified audio signals. The noise eliminating circuit includes a voltage detection circuit and a switch circuit; wherein when the voltage detection circuit detects a high voltage level power on/off signal in the audio signals received by the audio codec chip, the voltage detection circuit turns on the switch circuit. The audio amplifier chip is grounded via the switch circuit for eliminating interference of the noise signals.

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 boundary microphone, comprising: a base made of a metal; a cover made of a metal having a plurality of holes for introducing sound waves; and internal components including a microphone unit that is disposed in an internal space enclosed by the base and the cover; a first metallic part disposed on one side of the upper and lower sides of the internal components; and a second metallic part, which is disposed on other side of the upper and lower sides of the internal components and which encloses the internal components along with the first metallic part from all directions, are provided, wherein the base, the cover, the first metallic part, and the second metallic part are alternately overlapped at their peripheral portions, and wherein at least one of the first metallic part and the second metallic part is made of a metallic mesh.

Abstract: A video-audio recording and reproducing apparatus (101) has a built-in stereo microphone (21a, 21b) and an external microphone connection terminal (32). The external microphone connection terminal (32) is connected to a binaural microphone (3) to be attached to the ears of a photographer (300). When the binaural microphone (3) is used to collect ambient sounds, an audio signal to be recorded on a recording medium is switched from an audio signal from the built-in stereo microphone (21a, 21b) to a binaural audio signal from the binaural microphone (3). The photographer (300) puts the binaural microphone (3 (31a, 31b)) on his or her ears and collects ambient sounds around the photographer (300) including a sound emanating from an object. The object is photographed with a camera unit (11). The recording medium records the binaural audio signal, a photographed video signal, and a binaural flag signal.

Abstract: A signal processing apparatus includes an absolute value unit configured to convert an audio signal into absolute values, a representative value calculation unit configured to calculate representative values of consecutive sample values included in blocks of the audio signal which has been converted into the absolute values using at least maximum sample values among values of the samples included in the blocks for individual blocks, an average value calculation unit configured to determine a section which includes a predetermined number of consecutive blocks as a frame and calculate a maximum value of the representative values of the blocks included in the frame and an average value of the representative values of the blocks included in the frame, and a detector configured to detect click noise in the frame on the basis of a ratio of the maximum value to the average value.

Abstract: Methods and systems for selectively switching between microphones in a plurality of microphones are disclosed, including providing a first state that corresponds to one or more microphones selected from the plurality of microphones. A subset of microphones may be selected from the plurality of microphones in response to determining an average power of an input signal for each of the plurality of microphones. A second state may be identified that includes at least one of the subset of microphones in response to evaluating the average powers of the input signals for the subset of microphones against a predetermined condition. A transition from the first state to the second state may be delayed in response to determining a transition delay time corresponding to the first state.

Abstract: Methods and arrangements for controlling modes of audio devices according to user selectable features are disclosed. The system can receive a user selection for an alternate mode for audio device. A monitor can monitor audio levels in a physical area and the system can automatically switch an audio device from a current mode to the alternate mode when the monitored audio level exceeds a predetermined threshold. The alternate mode can provide a higher audio level or volume than the current mode. The alternate mode can also be a closed caption mode, a record mode, and a transcript mode to name a few. Thus, when the ambient sound level reaches a predetermined level many audio devices can change modes based on user configured instructions.

Abstract: A wireless microphone system is disclosed that uses a multiplicity of antennas to reduce or eliminate effects of radio frequency interference and intermodulation distortion. Signals from the multiplicity of antennas are selectively filtered, amplified and distributed as left and right diversity signals to microphone receivers for audio processing.

Abstract: A communication device contains multiple microphones that receive acoustic signals from a user and from the background. The acoustic signals from the user are enhanced using the background acoustic signals to reduce background noise. The enhanced signal are transmitted to an emergency network when an emergency call is made from the communication device. The raw signals are stored in the communication device for later retrieval or are transmitted simultaneously with the enhanced signals. The enhanced signals are transmitted using a circuit-switched voice mode while the raw signals are transmitted using a packet-switched voice mode.

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: A signal processing circuit includes a delaying unit that is configured to carry out delay processing on the basis of periodicity information synchronized with the periodicity of a periodic noise included in an input signal and a filter unit that is configured to receive the input signal and has a notch characteristic at a frequency f. The frequency f satisfies f=N/T, where N represents an integer equal to or greater than one and T represents a delaying time applied by the delaying unit.

Abstract: An apparatus and method of canceling a vocal component includes a first vocal canceling unit to delay each of the left and right channel input signals for a predetermined time and to feed-forward cross mix the delayed left and right channel signals with the left and right channel input signals, a sound stage widening unit to delay each of the left and right channel signals output from the first vocal canceling unit for a predetermined time and to feedback cross mix the signal of each delayed channel signals with the left and right channel signals, and a second vocal canceling unit to low-pass filter the left and right channel signals output from the sound stage widening unit and to mix the low-pass filtered left and right low frequency components and a high frequency component of the difference between the left and right channels to cancel the vocal component from an audio signal.

Abstract: A method and apparatus are provided for estimating the noise spectrum of an audio signal occupying a certain frequency range. The amplitude of the audio signal is measured at only a subset of the frequency range having a low expected signal to noise ratio at the time of measuring such that the measured amplitude is indicative of noise amplitude at that subset of the frequency range. A predefined noise spectral shape is then adapted by fitting to the noise amplitude so as to form the estimated noise spectrum. The noise spectrum so estimated is used to remove interference noise from the audio signal.

Abstract: A sampled digital audio signal is displayed on a spectrogram, in terms of frequency vs. time. An unwanted noise in the signal is visible in the spectrogram and the portion of the signal containing the unwanted noise can be selected using time and frequency constraints. An estimate for the signal within the selected portion is then interpolated on the basis of desired portions of the signal outside the time constraints defining the selected portion.

Abstract: Microphone is arranged at an evaluation point at the front seat; a signal for controlling vibration noise at this position is sent out from speaker at the front seat; secondary sound for canceling an influence of secondary sound at the front seat on the rear seat is sent out from speaker at the rear seat; microphone is arranged at an evaluation point at the rear seat; a signal for controlling vibration noise at this position is sent out from speaker; and secondary sound for canceling an influence of secondary sound at the rear seat on the front seat is sent out from speaker at the front seat.

Abstract: A method for delivering a message to a recipient in an environment with ambient noise includes the steps of recording the ambient noise in the environment at a certain time interval, analyzing the recorded ambient noise to obtain an average power Pnoise or a RMS amplitude Anoise of the ambient noise, providing a predetermined desired SNRdesired, calculating an average signal power Psignal or a RMS amplitude Asignal of the message to be delivered based on the Pnoise or Anoise and the desired SNRdesired, and adjusting a volume of the message to be delivered according to the Psignal or Asignal. Alternatively, the actual SNRactual will be computed and the message will be repeated if the SNRactual falls below the SNRmin. Systems for delivering a message to a recipient in an environment with ambient noise and computer-readable media having computer-executable instructions for carrying out the methods are also provided.

Abstract: A sound processing apparatus includes a target sound emphasizing unit configured to acquire a sound frequency component by emphasizing target sound in input sound in which the target sound and noise are included, a target sound suppressing unit configured to acquire a noise frequency component by suppressing the target sound in the input sound, a gain computing unit configured to compute a gain value to be multiplied by the sound frequency component using a gain function that provides a gain value and has a slope that are less than predetermined values when an energy ratio of the sound frequency component to the noise frequency component is less than or equal to a predetermined value, and a gain multiplier unit configured to multiply the sound frequency component by the gain value computed by the gain computing unit.

Abstract: A headset system which may be utilized for various forms of communication and multimedia. The headset system may include a number of earpieces which may be configured to change its operation state based on a position of the earpiece relative to a user's ear and a desired use. The earpiece may be further configured to provide noise cancellation based on its position and operational state.

Abstract: A device and method are provided that reduce interference between a wireless communication device and a speaker. Generally speaking, a microphone input is monitored for detecting noise created by the interference. If noise is detected, a power transmission level of the wireless device is reduced from a standard power transmission level.

Abstract: Noise effects in a signal for driving a plant are reduced by generating a reference signal from the error signal. A signal generator generates a reference signal for input to a finite impulse response (FIR) filter. The error signal is produced by differencing the transfer function output and a disturbance signal. The error signal is input to the signal generator and to a least mean square calculator. The reference signal is input to a copy of the transfer function that outputs a modified reference signal. The modified reference signal is input to least mean square calculator. An LMS signal that updates the filter coefficients to minimize the mean square error is calculated and the LMS signal and the reference signal are input to the FIR filter with the FIR filter being arranged to process the LMS signal and the reference signal to minimize the error signal.

Abstract: A noisy audio signal, with user input device noise, is received. Particular frames in the audio signal that are corrupted by user input device noise are identified and removed. The removed audio data is then reconstructed to obtain a clean audio signal.

Abstract: A system for correcting erroneous microphone readings in a vehicle engine harmonic cancellation (EHC) system. A method for operating an engine harmonic cancelling system, includes receiving, from a first microphone at a first location in a vehicle cabin, a signal representative of noise in the vehicle cabin; receiving, from a second microphone at a second location in the vehicle cabin, a signal representative of noise in the vehicle cabin; and correlating the signal from the first microphone with the signal from the second microphone.

Abstract: It is intended to improve and automate the calculation of calibration filters connected downstream from the microphones of an RGSC beamformer. To this end it is proposed that an adaptive calibration filter calculation unit be used, by means of which calibration filters are calculated from the output signals of adaptive blocking filters such that the power of an output signal of a blocking filter subtracted from a reference signal and filtered by means of a calibration filter respectively is minimized. The calibration filters connected downstream from the microphones are then replaced by the calibration filters thus determined.

Abstract: A signal processing apparatus includes a signal transforming unit transforming a first sound signal obtained by dividing a sound signal represented by a time function and by the predetermined time width into a second sound signal represented by a frequency function; a calculating unit determining a third sound signal which has a reduced influence of the operation sound by using the second sound signal and a sound signal representing the operation sound; a correcting unit performing a correction, by setting a sound signal representing the target sound as a reference signal, on the third sound signal based on the reference signal; and a signal inverse transforming unit inverse-transforming a sound signal on which the correction is performed from the sound signal represented by the frequency function into the sound signal represented by the time function.

Abstract: A system and method for combining multiple media types for demanding mobile communications environments, such as airborne and ship borne communications. The system facilitates integration of multiple media types into a common system, enabling use of the optimal communication system for each media type when possible, and transfer of media between communication systems as necessary. The system facilitates convergence of various media types to a common communications system, for example transferring voice and data over a common IP communications system, integrated control and management of the communications systems including interconnection configuration, priority, and accessibility with integrated management of shared resources. The system is designed to meet all Federal Aviation Administration (FAA) regulatory requirements.

Abstract: A novel adaptive beamforming technique with enhanced noise suppression capability. The technique incorporates the sound-source presence probability into an adaptive blocking matrix. In one embodiment the sound-source presence probability is estimated based on the instantaneous direction of arrival of the input signals and voice activity detection. The technique guarantees robustness to steering vector errors without imposing ad hoc constraints on the adaptive filter coefficients. It can provide good suppression performance for both directional interference signals as well as isotropic ambient noise.

Abstract: A system and a methods for correcting, simultaneously at multiple-listener positions, distortions introduced by the acoustical characteristics includes warping room responses, intelligently weighing the warped room acoustical responses to form a weighted response, a low order spectral fitting to the weighted response, forming a warped filter from the low order spectral fit, and unwarping the warped filter to form the room acoustical correction filter.